FFmpeg: libavcodec/alacenc.c Source File

FFmpeg

[フレーム]

libavcodec

alacenc.c

Go to the documentation of this file.

1 /*

2 * ALAC audio encoder

4 *

5 * This file is part of FFmpeg.

6 *

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

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

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

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

11 *

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

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

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

15 * Lesser General Public License for more details.

16 *

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

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

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

20 */

22 #include "avcodec.h"

23 #include "put_bits.h"

24 #include "internal.h"

25 #include "lpc.h"

26 #include "mathops.h"

27 #include "alac_data.h"

29 #define DEFAULT_FRAME_SIZE 4096

30 #define ALAC_EXTRADATA_SIZE 36

31 #define ALAC_FRAME_HEADER_SIZE 55

32 #define ALAC_FRAME_FOOTER_SIZE 3

34 #define ALAC_ESCAPE_CODE 0x1FF

35 #define ALAC_MAX_LPC_ORDER 30

36 #define DEFAULT_MAX_PRED_ORDER 6

37 #define DEFAULT_MIN_PRED_ORDER 4

38 #define ALAC_MAX_LPC_PRECISION 9

39 #define ALAC_MAX_LPC_SHIFT 9

41 #define ALAC_CHMODE_LEFT_RIGHT 0

42 #define ALAC_CHMODE_LEFT_SIDE 1

43 #define ALAC_CHMODE_RIGHT_SIDE 2

44 #define ALAC_CHMODE_MID_SIDE 3

46 typedef struct RiceContext {

47 int history_mult;

48 int initial_history;

49 int k_modifier;

50 int rice_modifier;

51 } RiceContext;

53 typedef struct AlacLPCContext {

54 int lpc_order;

55 int lpc_coeff[ALAC_MAX_LPC_ORDER+1];

56 int lpc_quant;

57 } AlacLPCContext;

59 typedef struct AlacEncodeContext {

60 int frame_size; /**< current frame size */

61 int verbatim; /**< current frame verbatim mode flag */

62 int compression_level;

63 int min_prediction_order;

64 int max_prediction_order;

65 int max_coded_frame_size;

66 int write_sample_size;

67 int extra_bits;

68 int32_t sample_buf[2][DEFAULT_FRAME_SIZE];

69 int32_t predictor_buf[DEFAULT_FRAME_SIZE];

70 int interlacing_shift;

71 int interlacing_leftweight;

72 PutBitContext pbctx;

73 RiceContext rc;

74 AlacLPCContext lpc[2];

75 LPCContext lpc_ctx;

76 AVCodecContext *avctx;

77 } AlacEncodeContext;

80 static void init_sample_buffers(AlacEncodeContext *s, int channels,

81 uint8_t const *samples[2])

82 {

83 int ch, i;

84 int shift = av_get_bytes_per_sample(s->avctx->sample_fmt) * 8 -

85 s->avctx->bits_per_raw_sample;

87 #define COPY_SAMPLES(type) do { \

88 for (ch = 0; ch < channels; ch++) { \

89 int32_t *bptr = s->sample_buf[ch]; \

90 const type *sptr = (const type *)samples[ch]; \

91 for (i = 0; i < s->frame_size; i++) \

92 bptr[i] = sptr[i] >> shift; \

93 } \

94 } while (0)

96 if (s->avctx->sample_fmt == AV_SAMPLE_FMT_S32P)

97 COPY_SAMPLES(int32_t);

98 else

99 COPY_SAMPLES(int16_t);

100 }

101

102 static void encode_scalar(AlacEncodeContext *s, int x,

103 int k, int write_sample_size)

104 {

105 int divisor, q, r;

106

107 k = FFMIN(k, s->rc.k_modifier);

108 divisor = (1<<k) - 1;

109 q = x / divisor;

110 r = x % divisor;

111

112 if (q > 8) {

113 // write escape code and sample value directly

114 put_bits(&s->pbctx, 9, ALAC_ESCAPE_CODE);

115 put_bits(&s->pbctx, write_sample_size, x);

116 } else {

117 if (q)

118 put_bits(&s->pbctx, q, (1<<q) - 1);

119 put_bits(&s->pbctx, 1, 0);

120

121 if (k != 1) {

122 if (r > 0)

123 put_bits(&s->pbctx, k, r+1);

124 else

125 put_bits(&s->pbctx, k-1, 0);

126 }

127 }

128 }

129

130 static void write_element_header(AlacEncodeContext *s,

131 enum AlacRawDataBlockType element,

132 int instance)

133 {

134 int encode_fs = 0;

135

136 if (s->frame_size < DEFAULT_FRAME_SIZE)

137 encode_fs = 1;

138

139 put_bits(&s->pbctx, 3, element); // element type

140 put_bits(&s->pbctx, 4, instance); // element instance

141 put_bits(&s->pbctx, 12, 0); // unused header bits

142 put_bits(&s->pbctx, 1, encode_fs); // Sample count is in the header

143 put_bits(&s->pbctx, 2, s->extra_bits >> 3); // Extra bytes (for 24-bit)

144 put_bits(&s->pbctx, 1, s->verbatim); // Audio block is verbatim

145 if (encode_fs)

146 put_bits32(&s->pbctx, s->frame_size); // No. of samples in the frame

147 }

148

149 static void calc_predictor_params(AlacEncodeContext *s, int ch)

150 {

151 int32_t coefs[MAX_LPC_ORDER][MAX_LPC_ORDER];

152 int shift[MAX_LPC_ORDER];

153 int opt_order;

154

155 if (s->compression_level == 1) {

156 s->lpc[ch].lpc_order = 6;

157 s->lpc[ch].lpc_quant = 6;

158 s->lpc[ch].lpc_coeff[0] = 160;

159 s->lpc[ch].lpc_coeff[1] = -190;

160 s->lpc[ch].lpc_coeff[2] = 170;

161 s->lpc[ch].lpc_coeff[3] = -130;

162 s->lpc[ch].lpc_coeff[4] = 80;

163 s->lpc[ch].lpc_coeff[5] = -25;

164 } else {

165 opt_order = ff_lpc_calc_coefs(&s->lpc_ctx, s->sample_buf[ch],

166 s->frame_size,

167 s->min_prediction_order,

168 s->max_prediction_order,

169 ALAC_MAX_LPC_PRECISION, coefs, shift,

170 FF_LPC_TYPE_LEVINSON, 0,

171 ORDER_METHOD_EST, ALAC_MAX_LPC_SHIFT, 1);

172

173 s->lpc[ch].lpc_order = opt_order;

174 s->lpc[ch].lpc_quant = shift[opt_order-1];

175 memcpy(s->lpc[ch].lpc_coeff, coefs[opt_order-1], opt_order*sizeof(int));

176 }

177 }

178

179 static int estimate_stereo_mode(int32_t *left_ch, int32_t *right_ch, int n)

180 {

181 int i, best;

182 int32_t lt, rt;

183 uint64_t sum[4];

184 uint64_t score[4];

185

186 /* calculate sum of 2nd order residual for each channel */

187 sum[0] = sum[1] = sum[2] = sum[3] = 0;

188 for (i = 2; i < n; i++) {

189 lt = left_ch[i] - 2 * left_ch[i - 1] + left_ch[i - 2];

190 rt = right_ch[i] - 2 * right_ch[i - 1] + right_ch[i - 2];

191 sum[2] += FFABS((lt + rt) >> 1);

192 sum[3] += FFABS(lt - rt);

193 sum[0] += FFABS(lt);

194 sum[1] += FFABS(rt);

195 }

196

197 /* calculate score for each mode */

198 score[0] = sum[0] + sum[1];

199 score[1] = sum[0] + sum[3];

200 score[2] = sum[1] + sum[3];

201 score[3] = sum[2] + sum[3];

202

203 /* return mode with lowest score */

204 best = 0;

205 for (i = 1; i < 4; i++) {

206 if (score[i] < score[best])

207 best = i;

208 }

209 return best;

210 }

211

212 static void alac_stereo_decorrelation(AlacEncodeContext *s)

213 {

214 int32_t *left = s->sample_buf[0], *right = s->sample_buf[1];

215 int i, mode, n = s->frame_size;

216 int32_t tmp;

217

218 mode = estimate_stereo_mode(left, right, n);

219

220 switch (mode) {

221 case ALAC_CHMODE_LEFT_RIGHT:

222 s->interlacing_leftweight = 0;

223 s->interlacing_shift = 0;

224 break;

225 case ALAC_CHMODE_LEFT_SIDE:

226 for (i = 0; i < n; i++)

227 right[i] = left[i] - right[i];

228 s->interlacing_leftweight = 1;

229 s->interlacing_shift = 0;

230 break;

231 case ALAC_CHMODE_RIGHT_SIDE:

232 for (i = 0; i < n; i++) {

233 tmp = right[i];

234 right[i] = left[i] - right[i];

235 left[i] = tmp + (right[i] >> 31);

236 }

237 s->interlacing_leftweight = 1;

238 s->interlacing_shift = 31;

239 break;

240 default:

241 for (i = 0; i < n; i++) {

242 tmp = left[i];

243 left[i] = (tmp + right[i]) >> 1;

244 right[i] = tmp - right[i];

245 }

246 s->interlacing_leftweight = 1;

247 s->interlacing_shift = 1;

248 break;

249 }

250 }

251

252 static void alac_linear_predictor(AlacEncodeContext *s, int ch)

253 {

254 int i;

255 AlacLPCContext lpc = s->lpc[ch];

256

257 if (lpc.lpc_order == 31) {

258 s->predictor_buf[0] = s->sample_buf[ch][0];

259

260 for (i = 1; i < s->frame_size; i++) {

261 s->predictor_buf[i] = s->sample_buf[ch][i ] -

262 s->sample_buf[ch][i - 1];

263 }

264

265 return;

266 }

267

268 // generalised linear predictor

269

270 if (lpc.lpc_order > 0) {

271 int32_t *samples = s->sample_buf[ch];

272 int32_t *residual = s->predictor_buf;

273

274 // generate warm-up samples

275 residual[0] = samples[0];

276 for (i = 1; i <= lpc.lpc_order; i++)

277 residual[i] = sign_extend(samples[i] - samples[i-1], s->write_sample_size);

278

279 // perform lpc on remaining samples

280 for (i = lpc.lpc_order + 1; i < s->frame_size; i++) {

281 int sum = 1 << (lpc.lpc_quant - 1), res_val, j;

282

283 for (j = 0; j < lpc.lpc_order; j++) {

284 sum += (samples[lpc.lpc_order-j] - samples[0]) *

285 lpc.lpc_coeff[j];

286 }

287

288 sum >>= lpc.lpc_quant;

289 sum += samples[0];

290 residual[i] = sign_extend(samples[lpc.lpc_order+1] - sum,

291 s->write_sample_size);

292 res_val = residual[i];

293

294 if (res_val) {

295 int index = lpc.lpc_order - 1;

296 int neg = (res_val < 0);

297

298 while (index >= 0 && (neg ? (res_val < 0) : (res_val > 0))) {

299 int val = samples[0] - samples[lpc.lpc_order - index];

300 int sign = (val ? FFSIGN(val) : 0);

301

302 if (neg)

303 sign *= -1;

304

305 lpc.lpc_coeff[index] -= sign;

306 val *= sign;

307 res_val -= (val >> lpc.lpc_quant) * (lpc.lpc_order - index);

308 index--;

309 }

310 }

311 samples++;

312 }

313 }

314 }

315

316 static void alac_entropy_coder(AlacEncodeContext *s)

317 {

318 unsigned int history = s->rc.initial_history;

319 int sign_modifier = 0, i, k;

320 int32_t *samples = s->predictor_buf;

321

322 for (i = 0; i < s->frame_size;) {

323 int x;

324

325 k = av_log2((history >> 9) + 3);

326

327 x = -2 * (*samples) -1;

328 x ^= x >> 31;

329

330 samples++;

331 i++;

332

333 encode_scalar(s, x - sign_modifier, k, s->write_sample_size);

334

335 history += x * s->rc.history_mult -

336 ((history * s->rc.history_mult) >> 9);

337

338 sign_modifier = 0;

339 if (x > 0xFFFF)

340 history = 0xFFFF;

341

342 if (history < 128 && i < s->frame_size) {

343 unsigned int block_size = 0;

344

345 k = 7 - av_log2(history) + ((history + 16) >> 6);

346

347 while (*samples == 0 && i < s->frame_size) {

348 samples++;

349 i++;

350 block_size++;

351 }

352 encode_scalar(s, block_size, k, 16);

353 sign_modifier = (block_size <= 0xFFFF);

354 history = 0;

355 }

356

357 }

358 }

359

360 static void write_element(AlacEncodeContext *s,

361 enum AlacRawDataBlockType element, int instance,

362 const uint8_t *samples0, const uint8_t *samples1)

363 {

364 uint8_t const *samples[2] = { samples0, samples1 };

365 int i, j, channels;

366 int prediction_type = 0;

367 PutBitContext *pb = &s->pbctx;

368

369 channels = element == TYPE_CPE ? 2 : 1;

370

371 if (s->verbatim) {

372 write_element_header(s, element, instance);

373 /* samples are channel-interleaved in verbatim mode */

374 if (s->avctx->sample_fmt == AV_SAMPLE_FMT_S32P) {

375 int shift = 32 - s->avctx->bits_per_raw_sample;

376 int32_t const *samples_s32[2] = { (const int32_t *)samples0,

377 (const int32_t *)samples1 };

378 for (i = 0; i < s->frame_size; i++)

379 for (j = 0; j < channels; j++)

380 put_sbits(pb, s->avctx->bits_per_raw_sample,

381 samples_s32[j][i] >> shift);

382 } else {

383 int16_t const *samples_s16[2] = { (const int16_t *)samples0,

384 (const int16_t *)samples1 };

385 for (i = 0; i < s->frame_size; i++)

386 for (j = 0; j < channels; j++)

387 put_sbits(pb, s->avctx->bits_per_raw_sample,

388 samples_s16[j][i]);

389 }

390 } else {

391 s->write_sample_size = s->avctx->bits_per_raw_sample - s->extra_bits +

392 channels - 1;

393

394 init_sample_buffers(s, channels, samples);

395 write_element_header(s, element, instance);

396

397 if (channels == 2)

398 alac_stereo_decorrelation(s);

399 else

400 s->interlacing_shift = s->interlacing_leftweight = 0;

401 put_bits(pb, 8, s->interlacing_shift);

402 put_bits(pb, 8, s->interlacing_leftweight);

403

404 for (i = 0; i < channels; i++) {

405 calc_predictor_params(s, i);

406

407 put_bits(pb, 4, prediction_type);

408 put_bits(pb, 4, s->lpc[i].lpc_quant);

409

410 put_bits(pb, 3, s->rc.rice_modifier);

411 put_bits(pb, 5, s->lpc[i].lpc_order);

412 // predictor coeff. table

413 for (j = 0; j < s->lpc[i].lpc_order; j++)

414 put_sbits(pb, 16, s->lpc[i].lpc_coeff[j]);

415 }

416

417 // write extra bits if needed

418 if (s->extra_bits) {

419 uint32_t mask = (1 << s->extra_bits) - 1;

420 for (i = 0; i < s->frame_size; i++) {

421 for (j = 0; j < channels; j++) {

422 put_bits(pb, s->extra_bits, s->sample_buf[j][i] & mask);

423 s->sample_buf[j][i] >>= s->extra_bits;

424 }

425 }

426 }

427

428 // apply lpc and entropy coding to audio samples

429 for (i = 0; i < channels; i++) {

430 alac_linear_predictor(s, i);

431

432 // TODO: determine when this will actually help. for now it's not used.

433 if (prediction_type == 15) {

434 // 2nd pass 1st order filter

435 for (j = s->frame_size - 1; j > 0; j--)

436 s->predictor_buf[j] -= s->predictor_buf[j - 1];

437 }

438 alac_entropy_coder(s);

439 }

440 }

441 }

442

443 static int write_frame(AlacEncodeContext *s, AVPacket *avpkt,

444 uint8_t * const *samples)

445 {

446 PutBitContext *pb = &s->pbctx;

447 const enum AlacRawDataBlockType *ch_elements = ff_alac_channel_elements[s->avctx->channels - 1];

448 const uint8_t *ch_map = ff_alac_channel_layout_offsets[s->avctx->channels - 1];

449 int ch, element, sce, cpe;

450

451 init_put_bits(pb, avpkt->data, avpkt->size);

452

453 ch = element = sce = cpe = 0;

454 while (ch < s->avctx->channels) {

455 if (ch_elements[element] == TYPE_CPE) {

456 write_element(s, TYPE_CPE, cpe, samples[ch_map[ch]],

457 samples[ch_map[ch + 1]]);

458 cpe++;

459 ch += 2;

460 } else {

461 write_element(s, TYPE_SCE, sce, samples[ch_map[ch]], NULL);

462 sce++;

463 ch++;

464 }

465 element++;

466 }

467

468 put_bits(pb, 3, TYPE_END);

469 flush_put_bits(pb);

470

471 return put_bits_count(pb) >> 3;

472 }

473

474 static av_always_inline int get_max_frame_size(int frame_size, int ch, int bps)

475 {

476 int header_bits = 23 + 32 * (frame_size < DEFAULT_FRAME_SIZE);

477 return FFALIGN(header_bits + bps * ch * frame_size + 3, 8) / 8;

478 }

479

480 static av_cold int alac_encode_close(AVCodecContext *avctx)

481 {

482 AlacEncodeContext *s = avctx->priv_data;

483 ff_lpc_end(&s->lpc_ctx);

484 av_freep(&avctx->extradata);

485 avctx->extradata_size = 0;

486 return 0;

487 }

488

489 static av_cold int alac_encode_init(AVCodecContext *avctx)

490 {

491 AlacEncodeContext *s = avctx->priv_data;

492 int ret;

493 uint8_t *alac_extradata;

494

495 avctx->frame_size = s->frame_size = DEFAULT_FRAME_SIZE;

496

497 if (avctx->sample_fmt == AV_SAMPLE_FMT_S32P) {

498 if (avctx->bits_per_raw_sample != 24)

499 av_log(avctx, AV_LOG_WARNING, "encoding as 24 bits-per-sample\n");

500 avctx->bits_per_raw_sample = 24;

501 } else {

502 avctx->bits_per_raw_sample = 16;

503 s->extra_bits = 0;

504 }

505

506 // Set default compression level

507 if (avctx->compression_level == FF_COMPRESSION_DEFAULT)

508 s->compression_level = 2;

509 else

510 s->compression_level = av_clip(avctx->compression_level, 0, 2);

511

512 // Initialize default Rice parameters

513 s->rc.history_mult = 40;

514 s->rc.initial_history = 10;

515 s->rc.k_modifier = 14;

516 s->rc.rice_modifier = 4;

517

518 s->max_coded_frame_size = get_max_frame_size(avctx->frame_size,

519 avctx->channels,

520 avctx->bits_per_raw_sample);

521

522 avctx->extradata = av_mallocz(ALAC_EXTRADATA_SIZE + FF_INPUT_BUFFER_PADDING_SIZE);

523 if (!avctx->extradata) {

524 ret = AVERROR(ENOMEM);

525 goto error;

526 }

527 avctx->extradata_size = ALAC_EXTRADATA_SIZE;

528

529 alac_extradata = avctx->extradata;

530 AV_WB32(alac_extradata, ALAC_EXTRADATA_SIZE);

531 AV_WB32(alac_extradata+4, MKBETAG('a','l','a','c'));

532 AV_WB32(alac_extradata+12, avctx->frame_size);

533 AV_WB8 (alac_extradata+17, avctx->bits_per_raw_sample);

534 AV_WB8 (alac_extradata+21, avctx->channels);

535 AV_WB32(alac_extradata+24, s->max_coded_frame_size);

536 AV_WB32(alac_extradata+28,

537 avctx->sample_rate * avctx->channels * avctx->bits_per_raw_sample); // average bitrate

538 AV_WB32(alac_extradata+32, avctx->sample_rate);

539

540 // Set relevant extradata fields

541 if (s->compression_level > 0) {

542 AV_WB8(alac_extradata+18, s->rc.history_mult);

543 AV_WB8(alac_extradata+19, s->rc.initial_history);

544 AV_WB8(alac_extradata+20, s->rc.k_modifier);

545 }

546

547 s->min_prediction_order = DEFAULT_MIN_PRED_ORDER;

548 if (avctx->min_prediction_order >= 0) {

549 if (avctx->min_prediction_order < MIN_LPC_ORDER ||

550 avctx->min_prediction_order > ALAC_MAX_LPC_ORDER) {

551 av_log(avctx, AV_LOG_ERROR, "invalid min prediction order: %d\n",

552 avctx->min_prediction_order);

553 ret = AVERROR(EINVAL);

554 goto error;

555 }

556

557 s->min_prediction_order = avctx->min_prediction_order;

558 }

559

560 s->max_prediction_order = DEFAULT_MAX_PRED_ORDER;

561 if (avctx->max_prediction_order >= 0) {

562 if (avctx->max_prediction_order < MIN_LPC_ORDER ||

563 avctx->max_prediction_order > ALAC_MAX_LPC_ORDER) {

564 av_log(avctx, AV_LOG_ERROR, "invalid max prediction order: %d\n",

565 avctx->max_prediction_order);

566 ret = AVERROR(EINVAL);

567 goto error;

568 }

569

570 s->max_prediction_order = avctx->max_prediction_order;

571 }

572

573 if (s->max_prediction_order < s->min_prediction_order) {

574 av_log(avctx, AV_LOG_ERROR,

575 "invalid prediction orders: min=%d max=%d\n",

576 s->min_prediction_order, s->max_prediction_order);

577 ret = AVERROR(EINVAL);

578 goto error;

579 }

580

581 s->avctx = avctx;

582

583 if ((ret = ff_lpc_init(&s->lpc_ctx, avctx->frame_size,

584 s->max_prediction_order,

585 FF_LPC_TYPE_LEVINSON)) < 0) {

586 goto error;

587 }

588

589 return 0;

590 error:

591 alac_encode_close(avctx);

592 return ret;

593 }

594

595 static int alac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,

596 const AVFrame *frame, int *got_packet_ptr)

597 {

598 AlacEncodeContext *s = avctx->priv_data;

599 int out_bytes, max_frame_size, ret;

600

601 s->frame_size = frame->nb_samples;

602

603 if (frame->nb_samples < DEFAULT_FRAME_SIZE)

604 max_frame_size = get_max_frame_size(s->frame_size, avctx->channels,

605 avctx->bits_per_raw_sample);

606 else

607 max_frame_size = s->max_coded_frame_size;

608

609 if ((ret = ff_alloc_packet2(avctx, avpkt, 2 * max_frame_size)) < 0)

610 return ret;

611

612 /* use verbatim mode for compression_level 0 */

613 if (s->compression_level) {

614 s->verbatim = 0;

615 s->extra_bits = avctx->bits_per_raw_sample - 16;

616 } else {

617 s->verbatim = 1;

618 s->extra_bits = 0;

619 }

620

621 out_bytes = write_frame(s, avpkt, frame->extended_data);

622

623 if (out_bytes > max_frame_size) {

624 /* frame too large. use verbatim mode */

625 s->verbatim = 1;

626 s->extra_bits = 0;

627 out_bytes = write_frame(s, avpkt, frame->extended_data);

628 }

629

630 avpkt->size = out_bytes;

631 *got_packet_ptr = 1;

632 return 0;

633 }

634

635 AVCodec ff_alac_encoder = {

636 .name = "alac",

637 .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),

638 .type = AVMEDIA_TYPE_AUDIO,

639 .id = AV_CODEC_ID_ALAC,

640 .priv_data_size = sizeof(AlacEncodeContext),

641 .init = alac_encode_init,

642 .encode2 = alac_encode_frame,

643 .close = alac_encode_close,

644 .capabilities = CODEC_CAP_SMALL_LAST_FRAME,

645 .channel_layouts = ff_alac_channel_layouts,

646 .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S32P,

647 AV_SAMPLE_FMT_S16P,

648 AV_SAMPLE_FMT_NONE },

649 };

Generated on Sat Jan 25 2014 19:51:44 for FFmpeg by doxygen 1.8.2