FFmpeg: libavcodec/aacenc.c Source File

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libavcodec

aacenc.c

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1 /*

2 * AAC encoder

4 *

5 * This file is part of FFmpeg.

6 *

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

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

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

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

11 *

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

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

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

15 * Lesser General Public License for more details.

16 *

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

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

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

20 */

22 /**

23 * @file

24 * AAC encoder

25 */

27 /***********************************

28 * TODOs:

29 * add sane pulse detection

30 * add temporal noise shaping

31 ***********************************/

33 #include "libavutil/float_dsp.h"

34 #include "libavutil/opt.h"

35 #include "avcodec.h"

36 #include "put_bits.h"

37 #include "internal.h"

38 #include "mpeg4audio.h"

39 #include "kbdwin.h"

40 #include "sinewin.h"

42 #include "aac.h"

43 #include "aactab.h"

44 #include "aacenc.h"

46 #include "psymodel.h"

48 #define AAC_MAX_CHANNELS 6

50 #define ERROR_IF(cond, ...) \

51 if (cond) { \

52 av_log(avctx, AV_LOG_ERROR, __VA_ARGS__); \

53 return AVERROR(EINVAL); \

54 }

56 float ff_aac_pow34sf_tab[428];

58 static const uint8_t swb_size_1024_96[] = {

59 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8,

60 12, 12, 12, 12, 12, 16, 16, 24, 28, 36, 44,

61 64, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64

62 };

64 static const uint8_t swb_size_1024_64[] = {

65 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8,

66 12, 12, 12, 16, 16, 16, 20, 24, 24, 28, 36,

67 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40

68 };

70 static const uint8_t swb_size_1024_48[] = {

71 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,

72 12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,

73 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32,

74 96

75 };

77 static const uint8_t swb_size_1024_32[] = {

78 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,

79 12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,

80 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32

81 };

83 static const uint8_t swb_size_1024_24[] = {

84 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,

85 12, 12, 12, 12, 16, 16, 16, 20, 20, 24, 24, 28, 28,

86 32, 36, 36, 40, 44, 48, 52, 52, 64, 64, 64, 64, 64

87 };

89 static const uint8_t swb_size_1024_16[] = {

90 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,

91 12, 12, 12, 12, 12, 12, 12, 12, 12, 16, 16, 16, 16, 20, 20, 20, 24, 24, 28, 28,

92 32, 36, 40, 40, 44, 48, 52, 56, 60, 64, 64, 64

93 };

95 static const uint8_t swb_size_1024_8[] = {

96 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12,

97 16, 16, 16, 16, 16, 16, 16, 20, 20, 20, 20, 24, 24, 24, 28, 28,

98 32, 36, 36, 40, 44, 48, 52, 56, 60, 64, 80

99 };

100

101 static const uint8_t *swb_size_1024[] = {

102 swb_size_1024_96, swb_size_1024_96, swb_size_1024_64,

103 swb_size_1024_48, swb_size_1024_48, swb_size_1024_32,

104 swb_size_1024_24, swb_size_1024_24, swb_size_1024_16,

105 swb_size_1024_16, swb_size_1024_16, swb_size_1024_8

106 };

107

108 static const uint8_t swb_size_128_96[] = {

109 4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 36

110 };

111

112 static const uint8_t swb_size_128_48[] = {

113 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 16

114 };

115

116 static const uint8_t swb_size_128_24[] = {

117 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 16, 16, 20

118 };

119

120 static const uint8_t swb_size_128_16[] = {

121 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 20

122 };

123

124 static const uint8_t swb_size_128_8[] = {

125 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 12, 16, 20, 20

126 };

127

128 static const uint8_t *swb_size_128[] = {

129 /* the last entry on the following row is swb_size_128_64 but is a

130 duplicate of swb_size_128_96 */

131 swb_size_128_96, swb_size_128_96, swb_size_128_96,

132 swb_size_128_48, swb_size_128_48, swb_size_128_48,

133 swb_size_128_24, swb_size_128_24, swb_size_128_16,

134 swb_size_128_16, swb_size_128_16, swb_size_128_8

135 };

136

137 /** default channel configurations */

138 static const uint8_t aac_chan_configs[6][5] = {

139 {1, TYPE_SCE}, // 1 channel - single channel element

140 {1, TYPE_CPE}, // 2 channels - channel pair

141 {2, TYPE_SCE, TYPE_CPE}, // 3 channels - center + stereo

142 {3, TYPE_SCE, TYPE_CPE, TYPE_SCE}, // 4 channels - front center + stereo + back center

143 {3, TYPE_SCE, TYPE_CPE, TYPE_CPE}, // 5 channels - front center + stereo + back stereo

144 {4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_LFE}, // 6 channels - front center + stereo + back stereo + LFE

145 };

146

147 /**

148 * Table to remap channels from libavcodec's default order to AAC order.

149 */

150 static const uint8_t aac_chan_maps[AAC_MAX_CHANNELS][AAC_MAX_CHANNELS] = {

151 { 0 },

152 { 0, 1 },

153 { 2, 0, 1 },

154 { 2, 0, 1, 3 },

155 { 2, 0, 1, 3, 4 },

156 { 2, 0, 1, 4, 5, 3 },

157 };

158

159 /**

160 * Make AAC audio config object.

161 * @see 1.6.2.1 "Syntax - AudioSpecificConfig"

162 */

163 static void put_audio_specific_config(AVCodecContext *avctx)

164 {

165 PutBitContext pb;

166 AACEncContext *s = avctx->priv_data;

167

168 init_put_bits(&pb, avctx->extradata, avctx->extradata_size*8);

169 put_bits(&pb, 5, 2); //object type - AAC-LC

170 put_bits(&pb, 4, s->samplerate_index); //sample rate index

171 put_bits(&pb, 4, s->channels);

172 //GASpecificConfig

173 put_bits(&pb, 1, 0); //frame length - 1024 samples

174 put_bits(&pb, 1, 0); //does not depend on core coder

175 put_bits(&pb, 1, 0); //is not extension

176

177 //Explicitly Mark SBR absent

178 put_bits(&pb, 11, 0x2b7); //sync extension

179 put_bits(&pb, 5, AOT_SBR);

180 put_bits(&pb, 1, 0);

181 flush_put_bits(&pb);

182 }

183

184 #define WINDOW_FUNC(type) \

185 static void apply_ ##type ##_window(AVFloatDSPContext *fdsp, \

186 SingleChannelElement *sce, \

187 const float *audio)

188

189 WINDOW_FUNC(only_long)

190 {

191 const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;

192 const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;

193 float *out = sce->ret_buf;

194

195 fdsp->vector_fmul (out, audio, lwindow, 1024);

196 fdsp->vector_fmul_reverse(out + 1024, audio + 1024, pwindow, 1024);

197 }

198

199 WINDOW_FUNC(long_start)

200 {

201 const float *lwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;

202 const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;

203 float *out = sce->ret_buf;

204

205 fdsp->vector_fmul(out, audio, lwindow, 1024);

206 memcpy(out + 1024, audio + 1024, sizeof(out[0]) * 448);

207 fdsp->vector_fmul_reverse(out + 1024 + 448, audio + 1024 + 448, swindow, 128);

208 memset(out + 1024 + 576, 0, sizeof(out[0]) * 448);

209 }

210

211 WINDOW_FUNC(long_stop)

212 {

213 const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;

214 const float *swindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;

215 float *out = sce->ret_buf;

216

217 memset(out, 0, sizeof(out[0]) * 448);

218 fdsp->vector_fmul(out + 448, audio + 448, swindow, 128);

219 memcpy(out + 576, audio + 576, sizeof(out[0]) * 448);

220 fdsp->vector_fmul_reverse(out + 1024, audio + 1024, lwindow, 1024);

221 }

222

223 WINDOW_FUNC(eight_short)

224 {

225 const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;

226 const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;

227 const float *in = audio + 448;

228 float *out = sce->ret_buf;

229 int w;

230

231 for (w = 0; w < 8; w++) {

232 fdsp->vector_fmul (out, in, w ? pwindow : swindow, 128);

233 out += 128;

234 in += 128;

235 fdsp->vector_fmul_reverse(out, in, swindow, 128);

236 out += 128;

237 }

238 }

239

240 static void (*const apply_window[4])(AVFloatDSPContext *fdsp,

241 SingleChannelElement *sce,

242 const float *audio) = {

243 [ONLY_LONG_SEQUENCE] = apply_only_long_window,

244 [LONG_START_SEQUENCE] = apply_long_start_window,

245 [EIGHT_SHORT_SEQUENCE] = apply_eight_short_window,

246 [LONG_STOP_SEQUENCE] = apply_long_stop_window

247 };

248

249 static void apply_window_and_mdct(AACEncContext *s, SingleChannelElement *sce,

250 float *audio)

251 {

252 int i;

253 float *output = sce->ret_buf;

254

255 apply_window[sce->ics.window_sequence[0]](&s->fdsp, sce, audio);

256

257 if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE)

258 s->mdct1024.mdct_calc(&s->mdct1024, sce->coeffs, output);

259 else

260 for (i = 0; i < 1024; i += 128)

261 s->mdct128.mdct_calc(&s->mdct128, sce->coeffs + i, output + i*2);

262 memcpy(audio, audio + 1024, sizeof(audio[0]) * 1024);

263 }

264

265 /**

266 * Encode ics_info element.

267 * @see Table 4.6 (syntax of ics_info)

268 */

269 static void put_ics_info(AACEncContext *s, IndividualChannelStream *info)

270 {

271 int w;

272

273 put_bits(&s->pb, 1, 0); // ics_reserved bit

274 put_bits(&s->pb, 2, info->window_sequence[0]);

275 put_bits(&s->pb, 1, info->use_kb_window[0]);

276 if (info->window_sequence[0] != EIGHT_SHORT_SEQUENCE) {

277 put_bits(&s->pb, 6, info->max_sfb);

278 put_bits(&s->pb, 1, 0); // no prediction

279 } else {

280 put_bits(&s->pb, 4, info->max_sfb);

281 for (w = 1; w < 8; w++)

282 put_bits(&s->pb, 1, !info->group_len[w]);

283 }

284 }

285

286 /**

287 * Encode MS data.

288 * @see 4.6.8.1 "Joint Coding - M/S Stereo"

289 */

290 static void encode_ms_info(PutBitContext *pb, ChannelElement *cpe)

291 {

292 int i, w;

293

294 put_bits(pb, 2, cpe->ms_mode);

295 if (cpe->ms_mode == 1)

296 for (w = 0; w < cpe->ch[0].ics.num_windows; w += cpe->ch[0].ics.group_len[w])

297 for (i = 0; i < cpe->ch[0].ics.max_sfb; i++)

298 put_bits(pb, 1, cpe->ms_mask[w*16 + i]);

299 }

300

301 /**

302 * Produce integer coefficients from scalefactors provided by the model.

303 */

304 static void adjust_frame_information(ChannelElement *cpe, int chans)

305 {

306 int i, w, w2, g, ch;

307 int start, maxsfb, cmaxsfb;

308

309 for (ch = 0; ch < chans; ch++) {

310 IndividualChannelStream *ics = &cpe->ch[ch].ics;

311 start = 0;

312 maxsfb = 0;

313 cpe->ch[ch].pulse.num_pulse = 0;

314 for (w = 0; w < ics->num_windows*16; w += 16) {

315 for (g = 0; g < ics->num_swb; g++) {

316 //apply M/S

317 if (cpe->common_window && !ch && cpe->ms_mask[w + g]) {

318 for (i = 0; i < ics->swb_sizes[g]; i++) {

319 cpe->ch[0].coeffs[start+i] = (cpe->ch[0].coeffs[start+i] + cpe->ch[1].coeffs[start+i]) / 2.0;

320 cpe->ch[1].coeffs[start+i] = cpe->ch[0].coeffs[start+i] - cpe->ch[1].coeffs[start+i];

321 }

322 }

323 start += ics->swb_sizes[g];

324 }

325 for (cmaxsfb = ics->num_swb; cmaxsfb > 0 && cpe->ch[ch].zeroes[w+cmaxsfb-1]; cmaxsfb--)

326 ;

327 maxsfb = FFMAX(maxsfb, cmaxsfb);

328 }

329 ics->max_sfb = maxsfb;

330

331 //adjust zero bands for window groups

332 for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {

333 for (g = 0; g < ics->max_sfb; g++) {

334 i = 1;

335 for (w2 = w; w2 < w + ics->group_len[w]; w2++) {

336 if (!cpe->ch[ch].zeroes[w2*16 + g]) {

337 i = 0;

338 break;

339 }

340 }

341 cpe->ch[ch].zeroes[w*16 + g] = i;

342 }

343 }

344 }

345

346 if (chans > 1 && cpe->common_window) {

347 IndividualChannelStream *ics0 = &cpe->ch[0].ics;

348 IndividualChannelStream *ics1 = &cpe->ch[1].ics;

349 int msc = 0;

350 ics0->max_sfb = FFMAX(ics0->max_sfb, ics1->max_sfb);

351 ics1->max_sfb = ics0->max_sfb;

352 for (w = 0; w < ics0->num_windows*16; w += 16)

353 for (i = 0; i < ics0->max_sfb; i++)

354 if (cpe->ms_mask[w+i])

355 msc++;

356 if (msc == 0 || ics0->max_sfb == 0)

357 cpe->ms_mode = 0;

358 else

359 cpe->ms_mode = msc < ics0->max_sfb * ics0->num_windows ? 1 : 2;

360 }

361 }

362

363 /**

364 * Encode scalefactor band coding type.

365 */

366 static void encode_band_info(AACEncContext *s, SingleChannelElement *sce)

367 {

368 int w;

369

370 for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w])

371 s->coder->encode_window_bands_info(s, sce, w, sce->ics.group_len[w], s->lambda);

372 }

373

374 /**

375 * Encode scalefactors.

376 */

377 static void encode_scale_factors(AVCodecContext *avctx, AACEncContext *s,

378 SingleChannelElement *sce)

379 {

380 int off = sce->sf_idx[0], diff;

381 int i, w;

382

383 for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {

384 for (i = 0; i < sce->ics.max_sfb; i++) {

385 if (!sce->zeroes[w*16 + i]) {

386 diff = sce->sf_idx[w*16 + i] - off + SCALE_DIFF_ZERO;

387 av_assert0(diff >= 0 && diff <= 120);

388 off = sce->sf_idx[w*16 + i];

389 put_bits(&s->pb, ff_aac_scalefactor_bits[diff], ff_aac_scalefactor_code[diff]);

390 }

391 }

392 }

393 }

394

395 /**

396 * Encode pulse data.

397 */

398 static void encode_pulses(AACEncContext *s, Pulse *pulse)

399 {

400 int i;

401

402 put_bits(&s->pb, 1, !!pulse->num_pulse);

403 if (!pulse->num_pulse)

404 return;

405

406 put_bits(&s->pb, 2, pulse->num_pulse - 1);

407 put_bits(&s->pb, 6, pulse->start);

408 for (i = 0; i < pulse->num_pulse; i++) {

409 put_bits(&s->pb, 5, pulse->pos[i]);

410 put_bits(&s->pb, 4, pulse->amp[i]);

411 }

412 }

413

414 /**

415 * Encode spectral coefficients processed by psychoacoustic model.

416 */

417 static void encode_spectral_coeffs(AACEncContext *s, SingleChannelElement *sce)

418 {

419 int start, i, w, w2;

420

421 for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {

422 start = 0;

423 for (i = 0; i < sce->ics.max_sfb; i++) {

424 if (sce->zeroes[w*16 + i]) {

425 start += sce->ics.swb_sizes[i];

426 continue;

427 }

428 for (w2 = w; w2 < w + sce->ics.group_len[w]; w2++)

429 s->coder->quantize_and_encode_band(s, &s->pb, sce->coeffs + start + w2*128,

430 sce->ics.swb_sizes[i],

431 sce->sf_idx[w*16 + i],

432 sce->band_type[w*16 + i],

433 s->lambda);

434 start += sce->ics.swb_sizes[i];

435 }

436 }

437 }

438

439 /**

440 * Encode one channel of audio data.

441 */

442 static int encode_individual_channel(AVCodecContext *avctx, AACEncContext *s,

443 SingleChannelElement *sce,

444 int common_window)

445 {

446 put_bits(&s->pb, 8, sce->sf_idx[0]);

447 if (!common_window)

448 put_ics_info(s, &sce->ics);

449 encode_band_info(s, sce);

450 encode_scale_factors(avctx, s, sce);

451 encode_pulses(s, &sce->pulse);

452 put_bits(&s->pb, 1, 0); //tns

453 put_bits(&s->pb, 1, 0); //ssr

454 encode_spectral_coeffs(s, sce);

455 return 0;

456 }

457

458 /**

459 * Write some auxiliary information about the created AAC file.

460 */

461 static void put_bitstream_info(AACEncContext *s, const char *name)

462 {

463 int i, namelen, padbits;

464

465 namelen = strlen(name) + 2;

466 put_bits(&s->pb, 3, TYPE_FIL);

467 put_bits(&s->pb, 4, FFMIN(namelen, 15));

468 if (namelen >= 15)

469 put_bits(&s->pb, 8, namelen - 14);

470 put_bits(&s->pb, 4, 0); //extension type - filler

471 padbits = -put_bits_count(&s->pb) & 7;

472 avpriv_align_put_bits(&s->pb);

473 for (i = 0; i < namelen - 2; i++)

474 put_bits(&s->pb, 8, name[i]);

475 put_bits(&s->pb, 12 - padbits, 0);

476 }

477

478 /*

479 * Copy input samples.

480 * Channels are reordered from libavcodec's default order to AAC order.

481 */

482 static void copy_input_samples(AACEncContext *s, const AVFrame *frame)

483 {

484 int ch;

485 int end = 2048 + (frame ? frame->nb_samples : 0);

486 const uint8_t *channel_map = aac_chan_maps[s->channels - 1];

487

488 /* copy and remap input samples */

489 for (ch = 0; ch < s->channels; ch++) {

490 /* copy last 1024 samples of previous frame to the start of the current frame */

491 memcpy(&s->planar_samples[ch][1024], &s->planar_samples[ch][2048], 1024 * sizeof(s->planar_samples[0][0]));

492

493 /* copy new samples and zero any remaining samples */

494 if (frame) {

495 memcpy(&s->planar_samples[ch][2048],

496 frame->extended_data[channel_map[ch]],

497 frame->nb_samples * sizeof(s->planar_samples[0][0]));

498 }

499 memset(&s->planar_samples[ch][end], 0,

500 (3072 - end) * sizeof(s->planar_samples[0][0]));

501 }

502 }

503

504 static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,

505 const AVFrame *frame, int *got_packet_ptr)

506 {

507 AACEncContext *s = avctx->priv_data;

508 float **samples = s->planar_samples, *samples2, *la, *overlap;

509 ChannelElement *cpe;

510 int i, ch, w, g, chans, tag, start_ch, ret;

511 int chan_el_counter[4];

512 FFPsyWindowInfo windows[AAC_MAX_CHANNELS];

513

514 if (s->last_frame == 2)

515 return 0;

516

517 /* add current frame to queue */

518 if (frame) {

519 if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)

520 return ret;

521 }

522

523 copy_input_samples(s, frame);

524 if (s->psypp)

525 ff_psy_preprocess(s->psypp, s->planar_samples, s->channels);

526

527 if (!avctx->frame_number)

528 return 0;

529

530 start_ch = 0;

531 for (i = 0; i < s->chan_map[0]; i++) {

532 FFPsyWindowInfo* wi = windows + start_ch;

533 tag = s->chan_map[i+1];

534 chans = tag == TYPE_CPE ? 2 : 1;

535 cpe = &s->cpe[i];

536 for (ch = 0; ch < chans; ch++) {

537 IndividualChannelStream *ics = &cpe->ch[ch].ics;

538 int cur_channel = start_ch + ch;

539 overlap = &samples[cur_channel][0];

540 samples2 = overlap + 1024;

541 la = samples2 + (448+64);

542 if (!frame)

543 la = NULL;

544 if (tag == TYPE_LFE) {

545 wi[ch].window_type[0] = ONLY_LONG_SEQUENCE;

546 wi[ch].window_shape = 0;

547 wi[ch].num_windows = 1;

548 wi[ch].grouping[0] = 1;

549

550 /* Only the lowest 12 coefficients are used in a LFE channel.

551 * The expression below results in only the bottom 8 coefficients

552 * being used for 11.025kHz to 16kHz sample rates.

553 */

554 ics->num_swb = s->samplerate_index >= 8 ? 1 : 3;

555 } else {

556 wi[ch] = s->psy.model->window(&s->psy, samples2, la, cur_channel,

557 ics->window_sequence[0]);

558 }

559 ics->window_sequence[1] = ics->window_sequence[0];

560 ics->window_sequence[0] = wi[ch].window_type[0];

561 ics->use_kb_window[1] = ics->use_kb_window[0];

562 ics->use_kb_window[0] = wi[ch].window_shape;

563 ics->num_windows = wi[ch].num_windows;

564 ics->swb_sizes = s->psy.bands [ics->num_windows == 8];

565 ics->num_swb = tag == TYPE_LFE ? ics->num_swb : s->psy.num_bands[ics->num_windows == 8];

566 for (w = 0; w < ics->num_windows; w++)

567 ics->group_len[w] = wi[ch].grouping[w];

568

569 apply_window_and_mdct(s, &cpe->ch[ch], overlap);

570 }

571 start_ch += chans;

572 }

573 if ((ret = ff_alloc_packet2(avctx, avpkt, 8192 * s->channels)) < 0)

574 return ret;

575 do {

576 int frame_bits;

577

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

579

580 if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & CODEC_FLAG_BITEXACT))

581 put_bitstream_info(s, LIBAVCODEC_IDENT);

582 start_ch = 0;

583 memset(chan_el_counter, 0, sizeof(chan_el_counter));

584 for (i = 0; i < s->chan_map[0]; i++) {

585 FFPsyWindowInfo* wi = windows + start_ch;

586 const float *coeffs[2];

587 tag = s->chan_map[i+1];

588 chans = tag == TYPE_CPE ? 2 : 1;

589 cpe = &s->cpe[i];

590 put_bits(&s->pb, 3, tag);

591 put_bits(&s->pb, 4, chan_el_counter[tag]++);

592 for (ch = 0; ch < chans; ch++)

593 coeffs[ch] = cpe->ch[ch].coeffs;

594 s->psy.model->analyze(&s->psy, start_ch, coeffs, wi);

595 for (ch = 0; ch < chans; ch++) {

596 s->cur_channel = start_ch + ch;

597 s->coder->search_for_quantizers(avctx, s, &cpe->ch[ch], s->lambda);

598 }

599 cpe->common_window = 0;

600 if (chans > 1

601 && wi[0].window_type[0] == wi[1].window_type[0]

602 && wi[0].window_shape == wi[1].window_shape) {

603

604 cpe->common_window = 1;

605 for (w = 0; w < wi[0].num_windows; w++) {

606 if (wi[0].grouping[w] != wi[1].grouping[w]) {

607 cpe->common_window = 0;

608 break;

609 }

610 }

611 }

612 s->cur_channel = start_ch;

613 if (s->options.stereo_mode && cpe->common_window) {

614 if (s->options.stereo_mode > 0) {

615 IndividualChannelStream *ics = &cpe->ch[0].ics;

616 for (w = 0; w < ics->num_windows; w += ics->group_len[w])

617 for (g = 0; g < ics->num_swb; g++)

618 cpe->ms_mask[w*16+g] = 1;

619 } else if (s->coder->search_for_ms) {

620 s->coder->search_for_ms(s, cpe, s->lambda);

621 }

622 }

623 adjust_frame_information(cpe, chans);

624 if (chans == 2) {

625 put_bits(&s->pb, 1, cpe->common_window);

626 if (cpe->common_window) {

627 put_ics_info(s, &cpe->ch[0].ics);

628 encode_ms_info(&s->pb, cpe);

629 }

630 }

631 for (ch = 0; ch < chans; ch++) {

632 s->cur_channel = start_ch + ch;

633 encode_individual_channel(avctx, s, &cpe->ch[ch], cpe->common_window);

634 }

635 start_ch += chans;

636 }

637

638 frame_bits = put_bits_count(&s->pb);

639 if (frame_bits <= 6144 * s->channels - 3) {

640 s->psy.bitres.bits = frame_bits / s->channels;

641 break;

642 }

643

644 s->lambda *= avctx->bit_rate * 1024.0f / avctx->sample_rate / frame_bits;

645

646 } while (1);

647

648 put_bits(&s->pb, 3, TYPE_END);

649 flush_put_bits(&s->pb);

650 avctx->frame_bits = put_bits_count(&s->pb);

651

652 // rate control stuff

653 if (!(avctx->flags & CODEC_FLAG_QSCALE)) {

654 float ratio = avctx->bit_rate * 1024.0f / avctx->sample_rate / avctx->frame_bits;

655 s->lambda *= ratio;

656 s->lambda = FFMIN(s->lambda, 65536.f);

657 }

658

659 if (!frame)

660 s->last_frame++;

661

662 ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,

663 &avpkt->duration);

664

665 avpkt->size = put_bits_count(&s->pb) >> 3;

666 *got_packet_ptr = 1;

667 return 0;

668 }

669

670 static av_cold int aac_encode_end(AVCodecContext *avctx)

671 {

672 AACEncContext *s = avctx->priv_data;

673

674 ff_mdct_end(&s->mdct1024);

675 ff_mdct_end(&s->mdct128);

676 ff_psy_end(&s->psy);

677 if (s->psypp)

678 ff_psy_preprocess_end(s->psypp);

679 av_freep(&s->buffer.samples);

680 av_freep(&s->cpe);

681 ff_af_queue_close(&s->afq);

682 return 0;

683 }

684

685 static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s)

686 {

687 int ret = 0;

688

689 avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);

690

691 // window init

692 ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);

693 ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);

694 ff_init_ff_sine_windows(10);

695 ff_init_ff_sine_windows(7);

696

697 if (ret = ff_mdct_init(&s->mdct1024, 11, 0, 32768.0))

698 return ret;

699 if (ret = ff_mdct_init(&s->mdct128, 8, 0, 32768.0))

700 return ret;

701

702 return 0;

703 }

704

705 static av_cold int alloc_buffers(AVCodecContext *avctx, AACEncContext *s)

706 {

707 int ch;

708 FF_ALLOCZ_OR_GOTO(avctx, s->buffer.samples, 3 * 1024 * s->channels * sizeof(s->buffer.samples[0]), alloc_fail);

709 FF_ALLOCZ_OR_GOTO(avctx, s->cpe, sizeof(ChannelElement) * s->chan_map[0], alloc_fail);

710 FF_ALLOCZ_OR_GOTO(avctx, avctx->extradata, 5 + FF_INPUT_BUFFER_PADDING_SIZE, alloc_fail);

711

712 for(ch = 0; ch < s->channels; ch++)

713 s->planar_samples[ch] = s->buffer.samples + 3 * 1024 * ch;

714

715 return 0;

716 alloc_fail:

717 return AVERROR(ENOMEM);

718 }

719

720 static av_cold int aac_encode_init(AVCodecContext *avctx)

721 {

722 AACEncContext *s = avctx->priv_data;

723 int i, ret = 0;

724 const uint8_t *sizes[2];

725 uint8_t grouping[AAC_MAX_CHANNELS];

726 int lengths[2];

727

728 avctx->frame_size = 1024;

729

730 for (i = 0; i < 16; i++)

731 if (avctx->sample_rate == avpriv_mpeg4audio_sample_rates[i])

732 break;

733

734 s->channels = avctx->channels;

735

736 ERROR_IF(i == 16,

737 "Unsupported sample rate %d\n", avctx->sample_rate);

738 ERROR_IF(s->channels > AAC_MAX_CHANNELS,

739 "Unsupported number of channels: %d\n", s->channels);

740 ERROR_IF(avctx->profile != FF_PROFILE_UNKNOWN && avctx->profile != FF_PROFILE_AAC_LOW,

741 "Unsupported profile %d\n", avctx->profile);

742 ERROR_IF(1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * s->channels,

743 "Too many bits per frame requested\n");

744

745 s->samplerate_index = i;

746

747 s->chan_map = aac_chan_configs[s->channels-1];

748

749 if (ret = dsp_init(avctx, s))

750 goto fail;

751

752 if (ret = alloc_buffers(avctx, s))

753 goto fail;

754

755 avctx->extradata_size = 5;

756 put_audio_specific_config(avctx);

757

758 sizes[0] = swb_size_1024[i];

759 sizes[1] = swb_size_128[i];

760 lengths[0] = ff_aac_num_swb_1024[i];

761 lengths[1] = ff_aac_num_swb_128[i];

762 for (i = 0; i < s->chan_map[0]; i++)

763 grouping[i] = s->chan_map[i + 1] == TYPE_CPE;

764 if (ret = ff_psy_init(&s->psy, avctx, 2, sizes, lengths, s->chan_map[0], grouping))

765 goto fail;

766 s->psypp = ff_psy_preprocess_init(avctx);

767 s->coder = &ff_aac_coders[s->options.aac_coder];

768

769 if (HAVE_MIPSDSPR1)

770 ff_aac_coder_init_mips(s);

771

772 s->lambda = avctx->global_quality ? avctx->global_quality : 120;

773

774 ff_aac_tableinit();

775

776 for (i = 0; i < 428; i++)

777 ff_aac_pow34sf_tab[i] = sqrt(ff_aac_pow2sf_tab[i] * sqrt(ff_aac_pow2sf_tab[i]));

778

779 avctx->delay = 1024;

780 ff_af_queue_init(avctx, &s->afq);

781

782 return 0;

783 fail:

784 aac_encode_end(avctx);

785 return ret;

786 }

787

788 #define AACENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM

789 static const AVOption aacenc_options[] = {

790 {"stereo_mode", "Stereo coding method", offsetof(AACEncContext, options.stereo_mode), AV_OPT_TYPE_INT, {.i64 = 0}, -1, 1, AACENC_FLAGS, "stereo_mode"},

791 {"auto", "Selected by the Encoder", 0, AV_OPT_TYPE_CONST, {.i64 = -1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},

792 {"ms_off", "Disable Mid/Side coding", 0, AV_OPT_TYPE_CONST, {.i64 = 0 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},

793 {"ms_force", "Force Mid/Side for the whole frame if possible", 0, AV_OPT_TYPE_CONST, {.i64 = 1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},

794 {"aac_coder", "", offsetof(AACEncContext, options.aac_coder), AV_OPT_TYPE_INT, {.i64 = 2}, 0, AAC_CODER_NB-1, AACENC_FLAGS},

795 {NULL}

796 };

797

798 static const AVClass aacenc_class = {

799 "AAC encoder",

800 av_default_item_name,

801 aacenc_options,

802 LIBAVUTIL_VERSION_INT,

803 };

804

805 /* duplicated from avpriv_mpeg4audio_sample_rates to avoid shared build

806 * failures */

807 static const int mpeg4audio_sample_rates[16] = {

808 96000, 88200, 64000, 48000, 44100, 32000,

809 24000, 22050, 16000, 12000, 11025, 8000, 7350

810 };

811

812 AVCodec ff_aac_encoder = {

813 .name = "aac",

814 .type = AVMEDIA_TYPE_AUDIO,

815 .id = AV_CODEC_ID_AAC,

816 .priv_data_size = sizeof(AACEncContext),

817 .init = aac_encode_init,

818 .encode2 = aac_encode_frame,

819 .close = aac_encode_end,

820 .supported_samplerates = mpeg4audio_sample_rates,

821 .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY |

822 CODEC_CAP_EXPERIMENTAL,

823 .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP,

824 AV_SAMPLE_FMT_NONE },

825 .long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),

826 .priv_class = &aacenc_class,

827 };

Generated on Wed Jul 10 2013 23:47:54 for FFmpeg by doxygen 1.8.2