FFmpeg: libavcodec/opusdec.c Source File

FFmpeg

libavcodec

opusdec.c

Go to the documentation of this file.

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

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 };

58

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 };

64

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

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 /**

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;

84

85 rc->range = 128;

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

87 rc->total_read_bits = 9;

88 opus_rc_normalize(rc);

89

90 return 0;

91 }

92

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 float **out, int out_size,

368 int nb_samples)

369 {

370 int output_samples = 0;

371 int flush_needed = 0;

372 int i, j, ret;

373

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

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

376 s->out_size = out_size;

377

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

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

380 if (buf) {

381 int64_t cur_samplerate;

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

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

384 } else {

385 flush_needed = !!s->delayed_samples;

386 }

387 }

388

389 if (!buf && !flush_needed)

390 return 0;

391

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

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

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

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

396 if (!s->out_dummy)

397 return AVERROR(ENOMEM);

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

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

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

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

402 }

403

404 /* flush the resampler if necessary */

405 if (flush_needed) {

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

407 if (ret < 0) {

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

409 return ret;

410 }

411 swr_close(s->swr);

412 output_samples += s->delayed_samples;

413 s->delayed_samples = 0;

414

415 if (!buf)

416 goto finish;

417 }

418

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

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

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

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

423

424 if (samples < 0) {

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

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

427 return samples;

428

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

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

431 samples = s->packet.frame_duration;

432 }

433 output_samples += samples;

434

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

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

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

438 }

439

440 finish:

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

442 s->out_size = 0;

443

444 return output_samples;

445 }

446

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

448 int *got_frame_ptr, AVPacket *avpkt)

449 {

450 OpusContext *c = avctx->priv_data;

451 AVFrame *frame = data;

452 const uint8_t *buf = avpkt->data;

453 int buf_size = avpkt->size;

454 int coded_samples = 0;

455 int decoded_samples = INT_MAX;

456 int delayed_samples = 0;

457 int i, ret;

458

459 /* calculate the number of delayed samples */

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

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

462 s->out[0] =

463 s->out[1] = NULL;

464 delayed_samples = FFMAX(delayed_samples,

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

466 }

467

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

469 if (buf) {

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

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

472 if (ret < 0) {

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

474 return ret;

475 }

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

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

478 }

479

480 frame->nb_samples = coded_samples + delayed_samples;

481

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

483 if (!frame->nb_samples) {

484 *got_frame_ptr = 0;

485 return 0;

486 }

487

488 /* setup the data buffers */

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

490 if (ret < 0)

491 return ret;

492 frame->nb_samples = 0;

493

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

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

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

497 if (!map->copy)

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

499 }

500

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

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

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

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

505

506 float sync_dummy[32];

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

508

509 if (!out[0])

510 out[0] = sync_dummy;

511 if (!out[1])

512 out[1] = sync_dummy;

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

514 return AVERROR_BUG;

515

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

517 if (ret < 0)

518 return ret;

519

520 if (out_dummy & 1)

521 out[0] = NULL;

522 else

523 out[0] += ret;

524 if (out_dummy & 2)

525 out[1] = NULL;

526 else

527 out[1] += ret;

528

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

530 }

531

532 /* decode each sub-packet */

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

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

535

536 if (i && buf) {

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

538 if (ret < 0) {

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

540 return ret;

541 }

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

543 av_log(avctx, AV_LOG_ERROR,

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

545 return AVERROR_INVALIDDATA;

546 }

547

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

549 }

550

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

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

553 if (ret < 0)

554 return ret;

555 c->decoded_samples[i] = ret;

556 decoded_samples = FFMIN(decoded_samples, ret);

557

558 buf += s->packet.packet_size;

559 buf_size -= s->packet.packet_size;

560 }

561

562 /* buffer the extra samples */

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

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

565 if (buffer_samples) {

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

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

568 buf[0] += decoded_samples;

569 buf[1] += decoded_samples;

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

571 if (ret < 0)

572 return ret;

573 }

574 }

575

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

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

578

579 /* handle copied channels */

580 if (map->copy) {

581 memcpy(frame->extended_data[i],

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

583 frame->linesize[0]);

584 } else if (map->silence) {

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

586 }

587

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

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

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

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

592 }

593 }

594

595 frame->nb_samples = decoded_samples;

596 *got_frame_ptr = !!decoded_samples;

597

598 return avpkt->size;

599 }

600

601 static av_cold void opus_decode_flush(AVCodecContext *ctx)

602 {

603 OpusContext *c = ctx->priv_data;

604 int i;

605

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

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

608

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

610 s->delayed_samples = 0;

611

612 if (s->celt_delay)

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

614 swr_close(s->swr);

615

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

617

618 ff_silk_flush(s->silk);

619 ff_celt_flush(s->celt);

620 }

621 }

622

623 static av_cold int opus_decode_close(AVCodecContext *avctx)

624 {

625 OpusContext *c = avctx->priv_data;

626 int i;

627

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

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

630

631 ff_silk_free(&s->silk);

632 ff_celt_free(&s->celt);

633

634 av_freep(&s->out_dummy);

635 s->out_dummy_allocated_size = 0;

636

637 av_audio_fifo_free(s->celt_delay);

638 swr_free(&s->swr);

639 }

640

641 av_freep(&c->streams);

642

643 if (c->sync_buffers) {

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

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

646 }

647 av_freep(&c->sync_buffers);

648 av_freep(&c->decoded_samples);

649 av_freep(&c->out);

650 av_freep(&c->out_size);

651

652 c->nb_streams = 0;

653

654 av_freep(&c->channel_maps);

655 av_freep(&c->fdsp);

656

657 return 0;

658 }

659

660 static av_cold int opus_decode_init(AVCodecContext *avctx)

661 {

662 OpusContext *c = avctx->priv_data;

663 int ret, i, j;

664

665 avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;

666 avctx->sample_rate = 48000;

667

668 c->fdsp = avpriv_float_dsp_alloc(0);

669 if (!c->fdsp)

670 return AVERROR(ENOMEM);

671

672 /* find out the channel configuration */

673 ret = ff_opus_parse_extradata(avctx, c);

674 if (ret < 0) {

675 av_freep(&c->channel_maps);

676 av_freep(&c->fdsp);

677 return ret;

678 }

679

680 /* allocate and init each independent decoder */

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

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

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

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

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

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

687 c->nb_streams = 0;

688 ret = AVERROR(ENOMEM);

689 goto fail;

690 }

691

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

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

694 uint64_t layout;

695

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

697

698 s->avctx = avctx;

699

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

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

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

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

704 }

705

706 s->fdsp = c->fdsp;

707

708 s->swr =swr_alloc();

709 if (!s->swr)

710 goto fail;

711

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

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

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

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

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

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

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

719

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

721 if (ret < 0)

722 goto fail;

723

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

725 if (ret < 0)

726 goto fail;

727

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

729 s->output_channels, 1024);

730 if (!s->celt_delay) {

731 ret = AVERROR(ENOMEM);

732 goto fail;

733 }

734

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

736 s->output_channels, 32);

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

738 ret = AVERROR(ENOMEM);

739 goto fail;

740 }

741 }

742

743 return 0;

744 fail:

745 opus_decode_close(avctx);

746 return ret;

747 }

748

749 AVCodec ff_opus_decoder = {

750 .name = "opus",

751 .long_name = NULL_IF_CONFIG_SMALL("Opus"),

752 .type = AVMEDIA_TYPE_AUDIO,

753 .id = AV_CODEC_ID_OPUS,

754 .priv_data_size = sizeof(OpusContext),

755 .init = opus_decode_init,

756 .close = opus_decode_close,

757 .decode = opus_decode_packet,

758 .flush = opus_decode_flush,

759 .capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_DELAY,

760 };

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:89

static const uint8_t celt_band_end[]

Definition: opusdec.c:65

opus_decode_close

static av_cold int opus_decode_close(AVCodecContext *avctx)

Definition: opusdec.c:623

void ff_celt_flush(CeltContext *s)

Definition: opus_celt.c:2147

AV_SAMPLE_FMT_FLTP

float, planar

Definition: samplefmt.h:70

#define NULL

Definition: coverity.c:32

s

const char * s

Definition: avisynth_c.h:631

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:60

av_cold void swr_close(SwrContext *s)

Closes the context so that swr_is_initialized() returns 0.

Definition: swresample.c:151

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:158

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

Definition: frame.h:181

OpusContext::sync_buffers

AVAudioFifo ** sync_buffers

Definition: opus.h:182

ptrdiff_t const GLvoid * data

Definition: opengl_enc.c:101

static void flush(AVCodecContext *avctx)

Definition: aacdec_template.c:496

silk_frame_duration_ms

static const uint16_t silk_frame_duration_ms[16]

Definition: opusdec.c:51

OpusPacket::frame_count

int frame_count

frame count

Definition: opus.h:114

OpusContext::nb_stereo_streams

int nb_stereo_streams

Definition: opus.h:187

AVFormatContext * ctx

Definition: movenc-test.c:48

OpusStreamContext::redundancy_buf

float redundancy_buf[2][960]

Definition: opus.h:137

static unsigned int get_bits(GetBitContext *s, int n)

Read 1-25 bits.

Definition: get_bits.h:260

static void opus_raw_init(OpusRangeCoder *rc, const uint8_t *rightend, unsigned int bytes)

Definition: opusdec.c:93

#define AV_LOG_WARNING

Something somehow does not look correct.

Definition: log.h:182

Definition: opus.h:74

static av_cold int init(AVCodecContext *avctx)

Definition: avrndec.c:35

static FFServerConfig config

Definition: ffserver.c:202

OpusStreamContext::output_channels

int output_channels

Definition: opus.h:124

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:151

OpusContext::gain

float gain

Definition: opus.h:191

opus_decode_redundancy

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

Definition: opusdec.c:179

int size

Definition: avcodec.h:1468

OpusRangeCoder::rb

RawBitsContext rb

Definition: opus.h:96

opus_rc_p2model

static av_always_inline unsigned int opus_rc_p2model(OpusRangeCoder *rc, unsigned int bits)

Definition: opus.h:234

OPUS_BANDWIDTH_MEDIUMBAND

Definition: opus.h:81

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:82

AVCodec.

Definition: avcodec.h:3392

Definition: opus.h:76

OpusContext::gain_i

int16_t gain_i

Definition: opus.h:190

Macro definitions for various function/variable attributes.

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

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:881

RawBitsContext::cacheval

unsigned int cacheval

Definition: opus.h:91

OpusContext::decoded_samples

int * decoded_samples

Definition: opus.h:184

opus_flush_resample

static int opus_flush_resample(OpusStreamContext *s, int nb_samples)

Definition: opusdec.c:111

AVCodecContext::sample_fmt

enum AVSampleFormat sample_fmt

audio sample format

Definition: avcodec.h:2295

uint8_t

Definition: audio_convert.c:194

av_cold struct SwrContext * swr_alloc(void)

Allocate SwrContext.

Definition: options.c:148

#define av_cold

Definition: attributes.h:82

#define av_assert2(cond)

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

Definition: avassert.h:63

AVOptions.

OpusRangeCoder::total_read_bits

unsigned int total_read_bits

Definition: opus.h:99

opus_decode_frame

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

Definition: opusdec.c:206

ChannelMap::copy

int copy

Definition: opus.h:166

OpusStreamContext::silk

SilkContext * silk

Definition: opus.h:128

static AVFrame * frame

Definition: demuxing_decoding.c:53

void ff_celt_free(CeltContext **s)

Definition: opus_celt.c:2174

OpusStreamContext::out_size

int out_size

Definition: opus.h:142

uint8_t * data

Definition: avcodec.h:1467

bitstream reader API header.

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

Definition: opusdec.c:102

ptrdiff_t size

Definition: opengl_enc.c:101

OpusStreamContext::silk_output

float * silk_output[2]

Definition: opus.h:133

#define FFALIGN(x, a)

Definition: macros.h:48

#define av_log(a,...)

Definition: tableprint_vlc.h:28

ff_celt_window2

const float ff_celt_window2[120]

Definition: opus_celt.c:467

OpusStreamContext

Definition: opus.h:122

opus_decode_init

static av_cold int opus_decode_init(AVCodecContext *avctx)

Definition: opusdec.c:660

OpusStreamContext::fdsp

AVFloatDSPContext * fdsp

Definition: opus.h:130

OpusContext::channel_maps

ChannelMap * channel_maps

Definition: opus.h:193

libswresample public header

OpusContext::nb_streams

int nb_streams

Definition: opus.h:186

#define AV_LOG_ERROR

Something went wrong and cannot losslessly be recovered.

Definition: log.h:176

#define AVERROR(e)

Definition: error.h:43

OpusRangeCoder::value

unsigned int value

Definition: opus.h:98

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:176

int out_size

Definition: movenc-test.c:55

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

Definition: opt.c:486

AVMEDIA_TYPE_AUDIO

Definition: avutil.h:194

OpusContext::fdsp

AVFloatDSPContext * fdsp

Definition: opus.h:189

const char * name

Name of the codec implementation.

Definition: avcodec.h:3399

#define FFMAX(a, b)

Definition: common.h:94

#define fail()

Definition: checkasm.h:80

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:205

int ff_celt_init(AVCodecContext *avctx, CeltContext **s, int output_channels)

Definition: opus_celt.c:2189

channel_layout.h

audio channel layout utility functions

OpusStreamContext::out

float * out[2]

Definition: opus.h:141

AVCodecContext::err_recognition

int err_recognition

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

Definition: avcodec.h:2811

OpusPacket::frame_size

int frame_size[MAX_FRAMES]

frame sizes

Definition: opus.h:116

#define FFMIN(a, b)

Definition: common.h:96

OpusPacket::frame_duration

int frame_duration

frame duration, in samples @ 48kHz

Definition: opus.h:117

OpusStreamContext::celt_buf

float celt_buf[2][960]

Definition: opus.h:134

OpusStreamContext::swr

SwrContext * swr

Definition: opus.h:147

OpusStreamContext::out_dummy_allocated_size

int out_dummy_allocated_size

Definition: opus.h:145

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:69

OpusStreamContext::silk_buf

float silk_buf[2][960]

Definition: opus.h:132

#define AV_EF_EXPLODE

abort decoding on minor error detection

Definition: avcodec.h:2822

AV_CODEC_ID_OPUS

Definition: avcodec.h:482

Definition: opus.h:106

#define FF_ARRAY_ELEMS(a)

Definition: sinewin_tablegen_template.c:36

Definition: opus.h:94

FILE * out

Definition: movenc-test.c:54

ChannelMap::silence

int silence

Definition: opus.h:171

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

Definition: opt.c:784

get_silk_samplerate

static int get_silk_samplerate(int config)

Definition: opusdec.c:67

Definition: opus.h:159

RawBitsContext::bytes

unsigned int bytes

Definition: opus.h:89

OpusStreamContext::out_dummy

float * out_dummy

Definition: opus.h:144

RawBitsContext::cachelen

unsigned int cachelen

Definition: opus.h:90

Libavcodec external API header.

OpusStreamContext::packet

OpusPacket packet

Definition: opus.h:153

AVCodecContext::sample_rate

int sample_rate

samples per second

Definition: avcodec.h:2287

AVFrame::linesize

int linesize[AV_NUM_DATA_POINTERS]

For video, size in bytes of each picture line.

Definition: frame.h:209

static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)

Initialize GetBitContext.

Definition: get_bits.h:449

ff_opus_decoder

AVCodec ff_opus_decoder

Definition: opusdec.c:749

static void finish(void)

Definition: movenc-test.c:299

void ff_silk_flush(SilkContext *s)

Definition: opus_silk.c:1567

main external API structure.

Definition: avcodec.h:1532

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

Definition: opus_silk.c:1575

av_cold void swr_free(SwrContext **ss)

Free the given SwrContext and set the pointer to NULL.

Definition: swresample.c:140

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

Get a buffer for a frame.

Definition: utils.c:894

OpusRangeCoder::gb

GetBitContext gb

Definition: opus.h:95

void * buf

Definition: avisynth_c.h:553

#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:112

void ff_silk_free(SilkContext **ps)

Definition: opus_silk.c:1562

OpusPacket::mode

enum OpusMode mode

mode

Definition: opus.h:118

ChannelMap::copy_idx

int copy_idx

Definition: opus.h:168

Definition: opus.h:174

avpriv_float_dsp_alloc

av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)

Allocate a float DSP context.

Definition: float_dsp.c:119

OpusPacket::stereo

int stereo

whether this packet is mono or stereo

Definition: opus.h:110

OpusStreamContext::avctx

AVCodecContext * avctx

Definition: opus.h:123

OpusContext::out

float ** out

Definition: opus.h:178

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:695

OpusPacket::data_size

int data_size

size of the useful data – packet size - padding

Definition: opus.h:108

ChannelMap::channel_idx

int channel_idx

Definition: opus.h:161

OpusStreamContext::celt

CeltContext * celt

Definition: opus.h:129

OpusStreamContext::redundancy_idx

int redundancy_idx

Definition: opus.h:155

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:113

opus_decode_packet

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

Definition: opusdec.c:447

av_audio_fifo_drain

int av_audio_fifo_drain(AVAudioFifo *af, int nb_samples)

Drain data from an AVAudioFifo.

Definition: audio_fifo.c:178

opus_init_resample

static int opus_init_resample(OpusStreamContext *s)

Definition: opusdec.c:154

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:365

static int decode(AVCodecContext *avctx, void *data, int *got_sub, AVPacket *avpkt)

Definition: ccaption_dec.c:572

OpusRangeCoder::range

unsigned int range

Definition: opus.h:97

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:499

opus_rc_normalize

static av_always_inline void opus_rc_normalize(OpusRangeCoder *rc)

Definition: opus.h:196

OpusStreamContext::celt_output

float * celt_output[2]

Definition: opus.h:135

common internal api header.

OpusStreamContext::rc

OpusRangeCoder rc

Definition: opus.h:126

ChannelMap::stream_idx

int stream_idx

Definition: opus.h:160

OpusContext::out_size

int * out_size

Definition: opus.h:179

OpusBandwidth

Definition: opus.h:79

c

static double c[64]

Definition: vsrc_mptestsrc.c:87

silk_resample_delay

static const int silk_resample_delay[]

Definition: opusdec.c:61

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:1498

opus_rc_unimodel

static av_always_inline unsigned int opus_rc_unimodel(OpusRangeCoder *rc, unsigned int size)

CELT: read a uniform distribution.

Definition: opus.h:303

ff_celt_decode_frame

int ff_celt_decode_frame(CeltContext *s, OpusRangeCoder *rc, float **output, int coded_channels, int frame_size, int startband, int endband)

Definition: opus_celt.c:1977

OpusContext::streams

OpusStreamContext * streams

Definition: opus.h:175

OpusPacket::packet_size

int packet_size

packet size

Definition: opus.h:107

OpusStreamContext::redundancy_rc

OpusRangeCoder redundancy_rc

Definition: opus.h:127

AVCodecContext::priv_data

void * priv_data

Definition: avcodec.h:1574

Audio FIFO Buffer.

int len

Definition: vorbis_enc_data.h:452

AVCodecContext::channels

int channels

number of audio channels

Definition: avcodec.h:2288

OpusPacket::frame_offset

int frame_offset[MAX_FRAMES]

frame offsets

Definition: opus.h:115

static av_always_inline unsigned int opus_rc_tell(const OpusRangeCoder *rc)

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

Definition: opus.h:255

OpusPacket::bandwidth

enum OpusBandwidth bandwidth

bandwidth

Definition: opus.h:119

opus_decode_flush

static av_cold void opus_decode_flush(AVCodecContext *ctx)

Definition: opusdec.c:601

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

Range decoder.

Definition: opusdec.c:79

OpusStreamContext::redundancy_output

float * redundancy_output[2]

Definition: opus.h:138

uint64_t layout

Definition: channel_layout.c:77

av_mallocz_array

static void * av_mallocz_array(size_t nmemb, size_t size)

Definition: mem.h:229

OpusStreamContext::celt_delay

AVAudioFifo * celt_delay

Definition: opus.h:148

ff_opus_parse_extradata

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

Definition: opus.c:290

#define av_freep(p)

Definition: tableprint_vlc.h:35

OpusStreamContext::silk_samplerate

int silk_samplerate

Definition: opus.h:149

RawBitsContext::position

const uint8_t * position

Definition: opus.h:88

AVFrame::extended_data

uint8_t ** extended_data

pointers to the data planes/channels.

Definition: frame.h:225

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:691

This structure stores compressed data.

Definition: avcodec.h:1444

OPUS_MODE_HYBRID

Definition: opus.h:75

AVFrame::nb_samples

int nb_samples

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

Definition: frame.h:235

AV_CODEC_CAP_DR1

#define AV_CODEC_CAP_DR1

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

Definition: avcodec.h:856

av_cold int swr_init(struct SwrContext *s)

Initialize context after user parameters have been set.

Definition: swresample.c:155

Generated on Mon Feb 15 2016 15:20:41 for FFmpeg by doxygen 1.8.6