FFmpeg: libavcodec/opusdec.c Source File

FFmpeg

libavcodec

opusdec.c

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1 /*

2 * Opus decoder

3 * Copyright (c) 2012 Andrew D'Addesio

4 * Copyright (c) 2013-2014 Mozilla Corporation

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 */

22

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 */

35

36 #include <stdint.h>

37

38 #include "libavutil/attributes.h"

39 #include "libavutil/audio_fifo.h"

40 #include "libavutil/channel_layout.h"

41 #include "libavutil/opt.h"

42

43 #include "libswresample/swresample.h"

44

45 #include "avcodec.h"

46 #include "get_bits.h"

47 #include "internal.h"

48 #include "mathops.h"

49 #include "opus.h"

50 #include "opustab.h"

51 #include "opus_celt.h"

52

53 static const uint16_t silk_frame_duration_ms[16] = {

54 10, 20, 40, 60,

55 10, 20, 40, 60,

56 10, 20, 40, 60,

57 10, 20,

58 10, 20,

59 };

60

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

62 * at the beginning */

63 static const int silk_resample_delay[] = {

64 4, 8, 11, 11, 11

65 };

66

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 }

75

76 static void opus_fade(float *out,

77 const float *in1, const float *in2,

78 const float *window, int len)

79 {

80 int i;

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

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

83 }

84

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

86 {

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

88 int ret, i;

89 ret = swr_convert(s->swr,

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

91 NULL, 0);

92 if (ret < 0)

93 return ret;

94 else if (ret != nb_samples) {

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

96 ret);

97 return AVERROR_BUG;

98 }

99

100 if (celt_size) {

101 if (celt_size != nb_samples) {

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

103 return AVERROR_BUG;

104 }

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

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

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

108 s->celt_output[i], 1.0,

109 nb_samples);

110 }

111 }

112

113 if (s->redundancy_idx) {

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

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

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

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

118 s->redundancy_idx = 0;

119 }

120

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

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

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

124

125 return 0;

126 }

127

128 static int opus_init_resample(OpusStreamContext *s)

129 {

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

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

132 int ret;

133

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

135 ret = swr_init(s->swr);

136 if (ret < 0) {

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

138 return ret;

139 }

140

141 ret = swr_convert(s->swr,

142 NULL, 0,

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

144 if (ret < 0) {

145 av_log(s->avctx, AV_LOG_ERROR,

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

147 return ret;

148 }

149

150 return 0;

151 }

152

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

154 {

155 int ret = ff_opus_rc_dec_init(&s->redundancy_rc, data, size);

156 if (ret < 0)

157 goto fail;

158 ff_opus_rc_dec_raw_init(&s->redundancy_rc, data + size, size);

159

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

161 s->redundancy_output,

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

163 0, ff_celt_band_end[s->packet.bandwidth]);

164 if (ret < 0)

165 goto fail;

166

167 return 0;

168 fail:

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

170 return ret;

171 }

172

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

174 {

175 int samples = s->packet.frame_duration;

176 int redundancy = 0;

177 int redundancy_size, redundancy_pos;

178 int ret, i, consumed;

179 int delayed_samples = s->delayed_samples;

180

181 ret = ff_opus_rc_dec_init(&s->rc, data, size);

182 if (ret < 0)

183 return ret;

184

185 /* decode the silk frame */

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

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

188 ret = opus_init_resample(s);

189 if (ret < 0)

190 return ret;

191 }

192

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

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

195 s->packet.stereo + 1,

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

197 if (samples < 0) {

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

199 return samples;

200 }

201 samples = swr_convert(s->swr,

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

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

204 if (samples < 0) {

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

206 return samples;

207 }

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

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

210 } else

211 ff_silk_flush(s->silk);

212

213 // decode redundancy information

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

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

216 redundancy = ff_opus_rc_dec_log(&s->rc, 12);

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

218 redundancy = 1;

219

220 if (redundancy) {

221 redundancy_pos = ff_opus_rc_dec_log(&s->rc, 1);

222

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

224 redundancy_size = ff_opus_rc_dec_uint(&s->rc, 256) + 2;

225 else

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

227 size -= redundancy_size;

228 if (size < 0) {

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

230 return AVERROR_INVALIDDATA;

231 }

232

233 if (redundancy_pos) {

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

235 if (ret < 0)

236 return ret;

237 ff_celt_flush(s->celt);

238 }

239 }

240

241 /* decode the CELT frame */

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

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

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

245 out_tmp : s->celt_output;

246 int celt_output_samples = samples;

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

248

249 if (delay_samples) {

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

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

252

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

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

255 delay_samples);

256 out_tmp[i] += delay_samples;

257 }

258 celt_output_samples -= delay_samples;

259 } else {

260 av_log(s->avctx, AV_LOG_WARNING,

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

262 av_audio_fifo_drain(s->celt_delay, delay_samples);

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

264 return AVERROR_BUG;

265 }

266 }

267

268 ff_opus_rc_dec_raw_init(&s->rc, data + size, size);

269

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

271 s->packet.stereo + 1,

272 s->packet.frame_duration,

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

274 ff_celt_band_end[s->packet.bandwidth]);

275 if (ret < 0)

276 return ret;

277

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

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

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

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

282

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

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

285 s->celt_output[i], 1.0,

286 celt_output_samples);

287 }

288

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

290 if (ret < 0)

291 return ret;

292 }

293 } else

294 ff_celt_flush(s->celt);

295

296 if (s->redundancy_idx) {

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

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

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

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

301 s->redundancy_idx = 0;

302 }

303 if (redundancy) {

304 if (!redundancy_pos) {

305 ff_celt_flush(s->celt);

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

307 if (ret < 0)

308 return ret;

309

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

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

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

313 s->redundancy_output[i] + 120,

314 ff_celt_window2, 120 - delayed_samples);

315 if (delayed_samples)

316 s->redundancy_idx = 120 - delayed_samples;

317 }

318 } else {

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

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

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

322 s->redundancy_output[i] + 120,

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

324 ff_celt_window2, 120);

325 }

326 }

327 }

328

329 return samples;

330 }

331

332 static int opus_decode_subpacket(OpusStreamContext *s,

333 const uint8_t *buf, int buf_size,

334 float **out, int out_size,

335 int nb_samples)

336 {

337 int output_samples = 0;

338 int flush_needed = 0;

339 int i, j, ret;

340

341 s->out[0] = out[0];

342 s->out[1] = out[1];

343 s->out_size = out_size;

344

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

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

347 if (buf) {

348 int64_t cur_samplerate;

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

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

351 } else {

352 flush_needed = !!s->delayed_samples;

353 }

354 }

355

356 if (!buf && !flush_needed)

357 return 0;

358

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

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

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

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

363 if (!s->out_dummy)

364 return AVERROR(ENOMEM);

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

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

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

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

369 }

370

371 /* flush the resampler if necessary */

372 if (flush_needed) {

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

374 if (ret < 0) {

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

376 return ret;

377 }

378 swr_close(s->swr);

379 output_samples += s->delayed_samples;

380 s->delayed_samples = 0;

381

382 if (!buf)

383 goto finish;

384 }

385

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

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

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

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

390

391 if (samples < 0) {

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

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

394 return samples;

395

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

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

398 samples = s->packet.frame_duration;

399 }

400 output_samples += samples;

401

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

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

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

405 }

406

407 finish:

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

409 s->out_size = 0;

410

411 return output_samples;

412 }

413

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

415 int *got_frame_ptr, AVPacket *avpkt)

416 {

417 OpusContext *c = avctx->priv_data;

418 AVFrame *frame = data;

419 const uint8_t *buf = avpkt->data;

420 int buf_size = avpkt->size;

421 int coded_samples = 0;

422 int decoded_samples = INT_MAX;

423 int delayed_samples = 0;

424 int i, ret;

425

426 /* calculate the number of delayed samples */

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

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

429 s->out[0] =

430 s->out[1] = NULL;

431 delayed_samples = FFMAX(delayed_samples,

432 s->delayed_samples + av_audio_fifo_size(c->sync_buffers[i]));

433 }

434

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

436 if (buf) {

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

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

439 if (ret < 0) {

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

441 return ret;

442 }

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

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

445 }

446

447 frame->nb_samples = coded_samples + delayed_samples;

448

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

450 if (!frame->nb_samples) {

451 *got_frame_ptr = 0;

452 return 0;

453 }

454

455 /* setup the data buffers */

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

457 if (ret < 0)

458 return ret;

459 frame->nb_samples = 0;

460

461 memset(c->out, 0, c->nb_streams * 2 * sizeof(*c->out));

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

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

464 if (!map->copy)

465 c->out[2 * map->stream_idx + map->channel_idx] = (float*)frame->extended_data[i];

466 }

467

468 /* read the data from the sync buffers */

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

470 float **out = c->out + 2 * i;

471 int sync_size = av_audio_fifo_size(c->sync_buffers[i]);

472

473 float sync_dummy[32];

474 int out_dummy = (!out[0]) | ((!out[1]) << 1);

475

476 if (!out[0])

477 out[0] = sync_dummy;

478 if (!out[1])

479 out[1] = sync_dummy;

480 if (out_dummy && sync_size > FF_ARRAY_ELEMS(sync_dummy))

481 return AVERROR_BUG;

482

483 ret = av_audio_fifo_read(c->sync_buffers[i], (void**)out, sync_size);

484 if (ret < 0)

485 return ret;

486

487 if (out_dummy & 1)

488 out[0] = NULL;

489 else

490 out[0] += ret;

491 if (out_dummy & 2)

492 out[1] = NULL;

493 else

494 out[1] += ret;

495

496 c->out_size[i] = frame->linesize[0] - ret * sizeof(float);

497 }

498

499 /* decode each sub-packet */

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

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

502

503 if (i && buf) {

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

505 if (ret < 0) {

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

507 return ret;

508 }

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

510 av_log(avctx, AV_LOG_ERROR,

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

512 return AVERROR_INVALIDDATA;

513 }

514

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

516 }

517

518 ret = opus_decode_subpacket(&c->streams[i], buf, s->packet.data_size,

519 c->out + 2 * i, c->out_size[i], coded_samples);

520 if (ret < 0)

521 return ret;

522 c->decoded_samples[i] = ret;

523 decoded_samples = FFMIN(decoded_samples, ret);

524

525 buf += s->packet.packet_size;

526 buf_size -= s->packet.packet_size;

527 }

528

529 /* buffer the extra samples */

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

531 int buffer_samples = c->decoded_samples[i] - decoded_samples;

532 if (buffer_samples) {

533 float *buf[2] = { c->out[2 * i + 0] ? c->out[2 * i + 0] : (float*)frame->extended_data[0],

534 c->out[2 * i + 1] ? c->out[2 * i + 1] : (float*)frame->extended_data[0] };

535 buf[0] += decoded_samples;

536 buf[1] += decoded_samples;

537 ret = av_audio_fifo_write(c->sync_buffers[i], (void**)buf, buffer_samples);

538 if (ret < 0)

539 return ret;

540 }

541 }

542

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

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

545

546 /* handle copied channels */

547 if (map->copy) {

548 memcpy(frame->extended_data[i],

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

550 frame->linesize[0]);

551 } else if (map->silence) {

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

553 }

554

555 if (c->gain_i && decoded_samples > 0) {

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

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

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

559 }

560 }

561

562 frame->nb_samples = decoded_samples;

563 *got_frame_ptr = !!decoded_samples;

564

565 return avpkt->size;

566 }

567

568 static av_cold void opus_decode_flush(AVCodecContext *ctx)

569 {

570 OpusContext *c = ctx->priv_data;

571 int i;

572

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

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

575

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

577 s->delayed_samples = 0;

578

579 if (s->celt_delay)

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

581 swr_close(s->swr);

582

583 av_audio_fifo_drain(c->sync_buffers[i], av_audio_fifo_size(c->sync_buffers[i]));

584

585 ff_silk_flush(s->silk);

586 ff_celt_flush(s->celt);

587 }

588 }

589

590 static av_cold int opus_decode_close(AVCodecContext *avctx)

591 {

592 OpusContext *c = avctx->priv_data;

593 int i;

594

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

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

597

598 ff_silk_free(&s->silk);

599 ff_celt_free(&s->celt);

600

601 av_freep(&s->out_dummy);

602 s->out_dummy_allocated_size = 0;

603

604 av_audio_fifo_free(s->celt_delay);

605 swr_free(&s->swr);

606 }

607

608 av_freep(&c->streams);

609

610 if (c->sync_buffers) {

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

612 av_audio_fifo_free(c->sync_buffers[i]);

613 }

614 av_freep(&c->sync_buffers);

615 av_freep(&c->decoded_samples);

616 av_freep(&c->out);

617 av_freep(&c->out_size);

618

619 c->nb_streams = 0;

620

621 av_freep(&c->channel_maps);

622 av_freep(&c->fdsp);

623

624 return 0;

625 }

626

627 static av_cold int opus_decode_init(AVCodecContext *avctx)

628 {

629 OpusContext *c = avctx->priv_data;

630 int ret, i, j;

631

632 avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;

633 avctx->sample_rate = 48000;

634

635 c->fdsp = avpriv_float_dsp_alloc(0);

636 if (!c->fdsp)

637 return AVERROR(ENOMEM);

638

639 /* find out the channel configuration */

640 ret = ff_opus_parse_extradata(avctx, c);

641 if (ret < 0) {

642 av_freep(&c->fdsp);

643 return ret;

644 }

645

646 /* allocate and init each independent decoder */

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

648 c->out = av_mallocz_array(c->nb_streams, 2 * sizeof(*c->out));

649 c->out_size = av_mallocz_array(c->nb_streams, sizeof(*c->out_size));

650 c->sync_buffers = av_mallocz_array(c->nb_streams, sizeof(*c->sync_buffers));

651 c->decoded_samples = av_mallocz_array(c->nb_streams, sizeof(*c->decoded_samples));

652 if (!c->streams || !c->sync_buffers || !c->decoded_samples || !c->out || !c->out_size) {

653 c->nb_streams = 0;

654 ret = AVERROR(ENOMEM);

655 goto fail;

656 }

657

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

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

660 uint64_t layout;

661

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

663

664 s->avctx = avctx;

665

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

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

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

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

670 }

671

672 s->fdsp = c->fdsp;

673

674 s->swr =swr_alloc();

675 if (!s->swr)

676 goto fail;

677

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

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

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

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

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

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

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

685

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

687 if (ret < 0)

688 goto fail;

689

690 ret = ff_celt_init(avctx, &s->celt, s->output_channels, c->apply_phase_inv);

691 if (ret < 0)

692 goto fail;

693

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

695 s->output_channels, 1024);

696 if (!s->celt_delay) {

697 ret = AVERROR(ENOMEM);

698 goto fail;

699 }

700

701 c->sync_buffers[i] = av_audio_fifo_alloc(avctx->sample_fmt,

702 s->output_channels, 32);

703 if (!c->sync_buffers[i]) {

704 ret = AVERROR(ENOMEM);

705 goto fail;

706 }

707 }

708

709 return 0;

710 fail:

711 opus_decode_close(avctx);

712 return ret;

713 }

714

715 #define OFFSET(x) offsetof(OpusContext, x)

716 #define AD AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_DECODING_PARAM

717 static const AVOption opus_options[] = {

718 { "apply_phase_inv", "Apply intensity stereo phase inversion", OFFSET(apply_phase_inv), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, AD },

719 { NULL },

720 };

721

722 static const AVClass opus_class = {

723 .class_name = "Opus Decoder",

724 .item_name = av_default_item_name,

725 .option = opus_options,

726 .version = LIBAVUTIL_VERSION_INT,

727 };

728

729 AVCodec ff_opus_decoder = {

730 .name = "opus",

731 .long_name = NULL_IF_CONFIG_SMALL("Opus"),

732 .priv_class = &opus_class,

733 .type = AVMEDIA_TYPE_AUDIO,

734 .id = AV_CODEC_ID_OPUS,

735 .priv_data_size = sizeof(OpusContext),

736 .init = opus_decode_init,

737 .close = opus_decode_close,

738 .decode = opus_decode_packet,

739 .flush = opus_decode_flush,

740 .capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_DELAY,

741 };

ff_opus_parse_packet

int ff_opus_parse_packet(OpusPacket *pkt, const uint8_t *buf, int buf_size, int self_delimiting)

Parse Opus packet info from raw packet data.

Definition: opus.c:90

opus_decode_close

static av_cold int opus_decode_close(AVCodecContext *avctx)

Definition: opusdec.c:590

AV_SAMPLE_FMT_FLTP

float, planar

Definition: samplefmt.h:69

#define NULL

Definition: coverity.c:32

ff_celt_decode_frame

int ff_celt_decode_frame(CeltFrame *f, OpusRangeCoder *rc, float **output, int channels, int frame_size, int start_band, int end_band)

Definition: opus_celt.c:347

s

const char * s

Definition: avisynth_c.h:768

AVERROR_INVALIDDATA

#define AVERROR_INVALIDDATA

Invalid data found when processing input.

Definition: error.h:59

av_audio_fifo_alloc

AVAudioFifo * av_audio_fifo_alloc(enum AVSampleFormat sample_fmt, int channels, int nb_samples)

Allocate an AVAudioFifo.

Definition: audio_fifo.c:59

av_cold void swr_close(SwrContext *s)

Closes the context so that swr_is_initialized() returns 0.

Definition: swresample.c:148

av_audio_fifo_read

int av_audio_fifo_read(AVAudioFifo *af, void **data, int nb_samples)

Read data from an AVAudioFifo.

Definition: audio_fifo.c:181

This structure describes decoded (raw) audio or video data.

Definition: frame.h:218

AVOption.

Definition: opt.h:246

OpusContext::sync_buffers

AVAudioFifo ** sync_buffers

Definition: opus.h:162

ptrdiff_t const GLvoid * data

Definition: opengl_enc.c:101

static void flush(AVCodecContext *avctx)

Definition: aacdec_template.c:500

silk_frame_duration_ms

static const uint16_t silk_frame_duration_ms[16]

Definition: opusdec.c:53

OpusPacket::frame_count

int frame_count

frame count

Definition: opus.h:92

OpusContext::nb_stereo_streams

int nb_stereo_streams

Definition: opus.h:167

OpusStreamContext::redundancy_buf

float redundancy_buf[2][960]

Definition: opus.h:115

#define AV_LOG_WARNING

Something somehow does not look correct.

Definition: log.h:182

Definition: opus.h:63

LIBAVUTIL_VERSION_INT

#define LIBAVUTIL_VERSION_INT

Definition: version.h:85

static av_cold int init(AVCodecContext *avctx)

Definition: avrndec.c:35

static const AVOption opus_options[]

Definition: opusdec.c:717

OpusStreamContext::output_channels

int output_channels

Definition: opus.h:102

av_audio_fifo_free

void av_audio_fifo_free(AVAudioFifo *af)

Free an AVAudioFifo.

Definition: audio_fifo.c:45

OpusStreamContext::delayed_samples

int delayed_samples

Definition: opus.h:129

OpusContext::gain

float gain

Definition: opus.h:171

opus_decode_redundancy

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

Definition: opusdec.c:153

int size

Definition: avcodec.h:1431

av_default_item_name

const char * av_default_item_name(void *ptr)

Return the context name.

Definition: log.c:191

#define OFFSET(x)

Definition: opusdec.c:715

int out_size

Definition: movenc.c:55

ff_opus_rc_dec_log

uint32_t ff_opus_rc_dec_log(OpusRangeCoder *rc, uint32_t bits)

Definition: opus_rc.c:114

static AVPacket pkt

Definition: demuxing_decoding.c:54

AV_CH_LAYOUT_STEREO

#define AV_CH_LAYOUT_STEREO

Definition: channel_layout.h:86

OPUS_BANDWIDTH_WIDEBAND

Definition: opus.h:73

AVCodec.

Definition: avcodec.h:3408

Definition: opus.h:65

static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)

Definition: decode_audio.c:42

OpusContext::gain_i

int16_t gain_i

Definition: opus.h:170

Macro definitions for various function/variable attributes.

ff_celt_band_end

const uint8_t ff_celt_band_end[]

Definition: opustab.c:27

AVFloatDSPContext::vector_fmac_scalar

void(* vector_fmac_scalar)(float *dst, const float *src, float mul, int len)

Multiply a vector of floats by a scalar float and add to destination vector.

Definition: float_dsp.h:54

OpusContext::apply_phase_inv

int apply_phase_inv

Definition: opus.h:155

AVClass::class_name

const char * class_name

The name of the class; usually it is the same name as the context structure type to which the AVClass...

Definition: log.h:72

AV_CODEC_CAP_DELAY

#define AV_CODEC_CAP_DELAY

Encoder or decoder requires flushing with NULL input at the end in order to give the complete and cor...

Definition: avcodec.h:984

OpusContext::decoded_samples

int * decoded_samples

Definition: opus.h:164

opus_flush_resample

static int opus_flush_resample(OpusStreamContext *s, int nb_samples)

Definition: opusdec.c:85

AVCodecContext::sample_fmt

enum AVSampleFormat sample_fmt

audio sample format

Definition: avcodec.h:2181

uint8_t

Definition: audio_convert.c:194

av_cold struct SwrContext * swr_alloc(void)

Allocate SwrContext.

Definition: options.c:149

#define av_cold

Definition: attributes.h:82

#define av_assert2(cond)

assert() equivalent, that does lie in speed critical code.

Definition: avassert.h:64

AVOptions.

opus_decode_frame

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

Definition: opusdec.c:173

ChannelMap::copy

int copy

Definition: opus.h:144

OpusStreamContext::silk

SilkContext * silk

Definition: opus.h:106

static AVFrame * frame

Definition: demuxing_decoding.c:53

static void finish(void)

Definition: movenc.c:345

OpusStreamContext::out_size

int out_size

Definition: opus.h:120

uint8_t * data

Definition: avcodec.h:1430

bitstream reader API header.

static void opus_fade(float *out, const float *in1, const float *in2, const float *window, int len)

Definition: opusdec.c:76

ptrdiff_t size

Definition: opengl_enc.c:101

OpusStreamContext::silk_output

float * silk_output[2]

Definition: opus.h:111

#define FFALIGN(x, a)

Definition: macros.h:48

#define av_log(a,...)

Definition: tableprint_vlc.h:28

OpusStreamContext

Definition: opus.h:100

opus_decode_init

static av_cold int opus_decode_init(AVCodecContext *avctx)

Definition: opusdec.c:627

OpusStreamContext::fdsp

AVFloatDSPContext * fdsp

Definition: opus.h:108

OpusContext::channel_maps

ChannelMap * channel_maps

Definition: opus.h:173

libswresample public header

OpusContext::nb_streams

int nb_streams

Definition: opus.h:166

#define AV_LOG_ERROR

Something went wrong and cannot losslessly be recovered.

Definition: log.h:176

avpriv_float_dsp_alloc

av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)

Allocate a float DSP context.

Definition: float_dsp.c:127

#define AVERROR(e)

Definition: error.h:43

NULL_IF_CONFIG_SMALL

#define NULL_IF_CONFIG_SMALL(x)

Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...

Definition: internal.h:186

void ff_celt_flush(CeltFrame *f)

Definition: opus_celt.c:522

int av_opt_set_int(void *obj, const char *name, int64_t val, int search_flags)

Definition: opt.c:558

AVMEDIA_TYPE_AUDIO

Definition: avutil.h:202

OpusContext::fdsp

AVFloatDSPContext * fdsp

Definition: opus.h:169

const char * name

Name of the codec implementation.

Definition: avcodec.h:3415

#define FFMAX(a, b)

Definition: common.h:94

#define fail()

Definition: checkasm.h:116

void av_fast_malloc(void *ptr, unsigned int *size, size_t min_size)

Allocate a buffer, reusing the given one if large enough.

Definition: mem.c:488

static SDL_Window * window

Definition: ffplay.c:362

av_audio_fifo_size

int av_audio_fifo_size(AVAudioFifo *af)

Get the current number of samples in the AVAudioFifo available for reading.

Definition: audio_fifo.c:228

channel_layout.h

audio channel layout utility functions

OpusStreamContext::out

float * out[2]

Definition: opus.h:119

AVCodecContext::err_recognition

int err_recognition

Error recognition; may misdetect some more or less valid parts as errors.

Definition: avcodec.h:2642

OpusPacket::frame_size

int frame_size[MAX_FRAMES]

frame sizes

Definition: opus.h:94

#define FFMIN(a, b)

Definition: common.h:96

OpusPacket::frame_duration

int frame_duration

frame duration, in samples @ 48kHz

Definition: opus.h:95

OpusStreamContext::celt_buf

float celt_buf[2][960]

Definition: opus.h:112

OpusStreamContext::swr

SwrContext * swr

Definition: opus.h:125

void ff_celt_free(CeltFrame **f)

Definition: opus_celt.c:549

AVFormatContext * ctx

Definition: movenc.c:48

AV_OPT_TYPE_BOOL

Definition: opt.h:240

OpusStreamContext::out_dummy_allocated_size

int out_dummy_allocated_size

Definition: opus.h:123

AVFloatDSPContext::vector_fmul_scalar

void(* vector_fmul_scalar)(float *dst, const float *src, float mul, int len)

Multiply a vector of floats by a scalar float.

Definition: float_dsp.h:85

OpusStreamContext::silk_buf

float silk_buf[2][960]

Definition: opus.h:110

#define AV_EF_EXPLODE

abort decoding on minor error detection

Definition: avcodec.h:2653

AV_CODEC_ID_OPUS

Definition: avcodec.h:611

Definition: opus.h:84

static const AVClass opus_class

Definition: opusdec.c:722

#define FF_ARRAY_ELEMS(a)

Definition: sinewin_tablegen_template.c:36

ChannelMap::silence

int silence

Definition: opus.h:149

int av_opt_get_int(void *obj, const char *name, int search_flags, int64_t *out_val)

Definition: opt.c:875

get_silk_samplerate

static int get_silk_samplerate(int config)

Definition: opusdec.c:67

Definition: opus.h:137

OpusStreamContext::celt

CeltFrame * celt

Definition: opus.h:107

OpusStreamContext::out_dummy

float * out_dummy

Definition: opus.h:122

Libavcodec external API header.

OpusStreamContext::packet

OpusPacket packet

Definition: opus.h:131

AVCodecContext::sample_rate

int sample_rate

samples per second

Definition: avcodec.h:2173

AVFrame::linesize

int linesize[AV_NUM_DATA_POINTERS]

For video, size in bytes of each picture line.

Definition: frame.h:249

ff_opus_decoder

AVCodec ff_opus_decoder

Definition: opusdec.c:729

void ff_silk_flush(SilkContext *s)

Definition: opus_silk.c:850

main external API structure.

Definition: avcodec.h:1518

ff_celt_window2

const float ff_celt_window2[120]

Definition: opustab.c:1133

int ff_silk_init(AVCodecContext *avctx, SilkContext **ps, int output_channels)

Definition: opus_silk.c:858

static av_always_inline uint32_t opus_rc_tell(const OpusRangeCoder *rc)

CELT: estimate bits of entropy that have thus far been consumed for the current CELT frame...

Definition: opus_rc.h:61

av_cold void swr_free(SwrContext **ss)

Free the given SwrContext and set the pointer to NULL.

Definition: swresample.c:137

int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)

Get a buffer for a frame.

Definition: decode.c:1891

void * buf

Definition: avisynth_c.h:690

#define AVERROR_BUG

Internal bug, also see AVERROR_BUG2.

Definition: error.h:50

OpusPacket::config

int config

configuration: tells the audio mode, bandwidth, and frame duration

Definition: opus.h:90

void ff_silk_free(SilkContext **ps)

Definition: opus_silk.c:845

Describe the class of an AVClass context structure.

Definition: log.h:67

OpusPacket::mode

enum OpusMode mode

mode

Definition: opus.h:96

ff_opus_rc_dec_raw_init

void ff_opus_rc_dec_raw_init(OpusRangeCoder *rc, const uint8_t *rightend, uint32_t bytes)

Definition: opus_rc.c:352

ChannelMap::copy_idx

int copy_idx

Definition: opus.h:146

Definition: opus.h:152

const VDPAUPixFmtMap * map

Definition: hwcontext_vdpau.c:83

OpusPacket::stereo

int stereo

whether this packet is mono or stereo

Definition: opus.h:88

OpusStreamContext::avctx

AVCodecContext * avctx

Definition: opus.h:101

OpusContext::out

float ** out

Definition: opus.h:158

int attribute_align_arg swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count, const uint8_t *in_arg[SWR_CH_MAX], int in_count)

Definition: swresample.c:706

OpusPacket::data_size

int data_size

size of the useful data – packet size - padding

Definition: opus.h:86

ChannelMap::channel_idx

int channel_idx

Definition: opus.h:139

OpusStreamContext::redundancy_idx

int redundancy_idx

Definition: opus.h:133

av_audio_fifo_write

int av_audio_fifo_write(AVAudioFifo *af, void **data, int nb_samples)

Write data to an AVAudioFifo.

Definition: audio_fifo.c:112

opus_decode_packet

static int opus_decode_packet(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)

Definition: opusdec.c:414

av_audio_fifo_drain

int av_audio_fifo_drain(AVAudioFifo *af, int nb_samples)

Drain data from an AVAudioFifo.

Definition: audio_fifo.c:201

opus_init_resample

static int opus_init_resample(OpusStreamContext *s)

Definition: opusdec.c:128

opus_decode_subpacket

static int opus_decode_subpacket(OpusStreamContext *s, const uint8_t *buf, int buf_size, float **out, int out_size, int nb_samples)

Definition: opusdec.c:332

OpusStreamContext::celt_output

float * celt_output[2]

Definition: opus.h:113

common internal api header.

OpusStreamContext::rc

OpusRangeCoder rc

Definition: opus.h:104

ChannelMap::stream_idx

int stream_idx

Definition: opus.h:138

OpusContext::out_size

int * out_size

Definition: opus.h:159

c

static double c[64]

Definition: vsrc_mptestsrc.c:87

silk_resample_delay

static const int silk_resample_delay[]

Definition: opusdec.c:63

ff_silk_decode_superframe

int ff_silk_decode_superframe(SilkContext *s, OpusRangeCoder *rc, float *output[2], enum OpusBandwidth bandwidth, int coded_channels, int duration_ms)

Decode the LP layer of one Opus frame (which may correspond to several SILK frames).

Definition: opus_silk.c:781

OpusContext::streams

OpusStreamContext * streams

Definition: opus.h:154

OpusPacket::packet_size

int packet_size

packet size

Definition: opus.h:85

OpusStreamContext::redundancy_rc

OpusRangeCoder redundancy_rc

Definition: opus.h:105

AVCodecContext::priv_data

void * priv_data

Definition: avcodec.h:1545

ff_opus_rc_dec_init

int ff_opus_rc_dec_init(OpusRangeCoder *rc, const uint8_t *data, int size)

Definition: opus_rc.c:338

Audio FIFO Buffer.

int len

Definition: vorbis_enc_data.h:452

AVCodecContext::channels

int channels

number of audio channels

Definition: avcodec.h:2174

OpusPacket::frame_offset

int frame_offset[MAX_FRAMES]

frame offsets

Definition: opus.h:93

OpusPacket::bandwidth

enum OpusBandwidth bandwidth

bandwidth

Definition: opus.h:97

opus_decode_flush

static av_cold void opus_decode_flush(AVCodecContext *ctx)

Definition: opusdec.c:568

OpusStreamContext::redundancy_output

float * redundancy_output[2]

Definition: opus.h:116

uint64_t layout

Definition: channel_layout.c:77

ff_opus_rc_dec_uint

uint32_t ff_opus_rc_dec_uint(OpusRangeCoder *rc, uint32_t size)

CELT: read a uniform distribution.

Definition: opus_rc.c:182

OpusStreamContext::celt_delay

AVAudioFifo * celt_delay

Definition: opus.h:126

ff_opus_parse_extradata

av_cold int ff_opus_parse_extradata(AVCodecContext *avctx, OpusContext *s)

Definition: opus.c:291

FILE * out

Definition: movenc.c:54

#define av_freep(p)

Definition: tableprint_vlc.h:35

OpusStreamContext::silk_samplerate

int silk_samplerate

Definition: opus.h:127

#define AD

Definition: opusdec.c:716

AVFrame::extended_data

uint8_t ** extended_data

pointers to the data planes/channels.

Definition: frame.h:265

AV_CH_LAYOUT_MONO

#define AV_CH_LAYOUT_MONO

Definition: channel_layout.h:85

swr_is_initialized

int swr_is_initialized(struct SwrContext *s)

Check whether an swr context has been initialized or not.

Definition: swresample.c:702

This structure stores compressed data.

Definition: avcodec.h:1407

OPUS_MODE_HYBRID

Definition: opus.h:64

AVFrame::nb_samples

int nb_samples

number of audio samples (per channel) described by this frame

Definition: frame.h:284

AV_CODEC_CAP_DR1

#define AV_CODEC_CAP_DR1

Codec uses get_buffer() for allocating buffers and supports custom allocators.

Definition: avcodec.h:959

int ff_celt_init(AVCodecContext *avctx, CeltFrame **f, int output_channels, int apply_phase_inv)

Definition: opus_celt.c:566

av_cold int swr_init(struct SwrContext *s)

Initialize context after user parameters have been set.

Definition: swresample.c:152

av_mallocz_array

void * av_mallocz_array(size_t nmemb, size_t size)

Definition: mem.c:191

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