FFmpeg: libavcodec/dcaenc.c Source File

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libavcodec

dcaenc.c

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1 /*

2 * DCA encoder

4 * 2010 Benjamin Larsson

5 * 2011 Xiang Wang

6 *

7 * This file is part of FFmpeg.

8 *

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

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

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

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

13 *

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

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

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

17 * Lesser General Public License for more details.

18 *

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

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

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

22 */

24 #include "libavutil/avassert.h"

25 #include "libavutil/channel_layout.h"

26 #include "libavutil/common.h"

27 #include "avcodec.h"

28 #include "dca.h"

29 #include "dcadata.h"

30 #include "dcaenc.h"

31 #include "internal.h"

32 #include "put_bits.h"

34 #define MAX_CHANNELS 6

35 #define DCA_MAX_FRAME_SIZE 16384

36 #define DCA_HEADER_SIZE 13

37 #define DCA_LFE_SAMPLES 8

39 #define DCA_SUBBANDS 32

40 #define SUBFRAMES 1

41 #define SUBSUBFRAMES 2

42 #define SUBBAND_SAMPLES (SUBFRAMES * SUBSUBFRAMES * 8)

43 #define AUBANDS 25

45 typedef struct DCAContext {

46 PutBitContext pb;

47 int frame_size;

48 int frame_bits;

49 int fullband_channels;

50 int channels;

51 int lfe_channel;

52 int samplerate_index;

53 int bitrate_index;

54 int channel_config;

55 const int32_t *band_interpolation;

56 const int32_t *band_spectrum;

57 int lfe_scale_factor;

58 softfloat lfe_quant;

59 int32_t lfe_peak_cb;

61 int32_t history[512][MAX_CHANNELS]; /* This is a circular buffer */

62 int32_t subband[SUBBAND_SAMPLES][DCA_SUBBANDS][MAX_CHANNELS];

63 int32_t quantized[SUBBAND_SAMPLES][DCA_SUBBANDS][MAX_CHANNELS];

64 int32_t peak_cb[DCA_SUBBANDS][MAX_CHANNELS];

65 int32_t downsampled_lfe[DCA_LFE_SAMPLES];

66 int32_t masking_curve_cb[SUBSUBFRAMES][256];

67 int abits[DCA_SUBBANDS][MAX_CHANNELS];

68 int scale_factor[DCA_SUBBANDS][MAX_CHANNELS];

69 softfloat quant[DCA_SUBBANDS][MAX_CHANNELS];

70 int32_t eff_masking_curve_cb[256];

71 int32_t band_masking_cb[32];

72 int32_t worst_quantization_noise;

73 int32_t worst_noise_ever;

74 int consumed_bits;

75 } DCAContext;

77 static int32_t cos_table[2048];

78 static int32_t band_interpolation[2][512];

79 static int32_t band_spectrum[2][8];

80 static int32_t auf[9][AUBANDS][256];

81 static int32_t cb_to_add[256];

82 static int32_t cb_to_level[2048];

83 static int32_t lfe_fir_64i[512];

85 /* Transfer function of outer and middle ear, Hz -> dB */

86 static double hom(double f)

87 {

88 double f1 = f / 1000;

90 return -3.64 * pow(f1, -0.8)

91 + 6.8 * exp(-0.6 * (f1 - 3.4) * (f1 - 3.4))

92 - 6.0 * exp(-0.15 * (f1 - 8.7) * (f1 - 8.7))

93 - 0.0006 * (f1 * f1) * (f1 * f1);

94 }

96 static double gammafilter(int i, double f)

97 {

98 double h = (f - fc[i]) / erb[i];

100 h = 1 + h * h;

101 h = 1 / (h * h);

102 return 20 * log10(h);

103 }

104

105 static int encode_init(AVCodecContext *avctx)

106 {

107 DCAContext *c = avctx->priv_data;

108 uint64_t layout = avctx->channel_layout;

109 int i, min_frame_bits;

110

111 c->fullband_channels = c->channels = avctx->channels;

112 c->lfe_channel = (avctx->channels == 3 || avctx->channels == 6);

113 c->band_interpolation = band_interpolation[1];

114 c->band_spectrum = band_spectrum[1];

115 c->worst_quantization_noise = -2047;

116 c->worst_noise_ever = -2047;

117

118 if (!layout) {

119 av_log(avctx, AV_LOG_WARNING, "No channel layout specified. The "

120 "encoder will guess the layout, but it "

121 "might be incorrect.\n");

122 layout = av_get_default_channel_layout(avctx->channels);

123 }

124 switch (layout) {

125 case AV_CH_LAYOUT_MONO: c->channel_config = 0; break;

126 case AV_CH_LAYOUT_STEREO: c->channel_config = 2; break;

127 case AV_CH_LAYOUT_2_2: c->channel_config = 8; break;

128 case AV_CH_LAYOUT_5POINT0: c->channel_config = 9; break;

129 case AV_CH_LAYOUT_5POINT1: c->channel_config = 9; break;

130 default:

131 av_log(avctx, AV_LOG_ERROR, "Unsupported channel layout!\n");

132 return AVERROR_PATCHWELCOME;

133 }

134

135 if (c->lfe_channel)

136 c->fullband_channels--;

137

138 for (i = 0; i < 9; i++) {

139 if (sample_rates[i] == avctx->sample_rate)

140 break;

141 }

142 if (i == 9)

143 return AVERROR(EINVAL);

144 c->samplerate_index = i;

145

146 if (avctx->bit_rate < 32000 || avctx->bit_rate > 3840000) {

147 av_log(avctx, AV_LOG_ERROR, "Bit rate %i not supported.", avctx->bit_rate);

148 return AVERROR(EINVAL);

149 }

150 for (i = 0; dca_bit_rates[i] < avctx->bit_rate; i++)

151 ;

152 c->bitrate_index = i;

153 avctx->bit_rate = dca_bit_rates[i];

154 c->frame_bits = FFALIGN((avctx->bit_rate * 512 + avctx->sample_rate - 1) / avctx->sample_rate, 32);

155 min_frame_bits = 132 + (493 + 28 * 32) * c->fullband_channels + c->lfe_channel * 72;

156 if (c->frame_bits < min_frame_bits || c->frame_bits > (DCA_MAX_FRAME_SIZE << 3))

157 return AVERROR(EINVAL);

158

159 c->frame_size = (c->frame_bits + 7) / 8;

160

161 avctx->frame_size = 32 * SUBBAND_SAMPLES;

162

163 if (!cos_table[0]) {

164 int j, k;

165

166 for (i = 0; i < 2048; i++) {

167 cos_table[i] = (int32_t)(0x7fffffff * cos(M_PI * i / 1024));

168 cb_to_level[i] = (int32_t)(0x7fffffff * pow(10, -0.005 * i));

169 }

170

171 /* FIXME: probably incorrect */

172 for (i = 0; i < 256; i++) {

173 lfe_fir_64i[i] = (int32_t)(0x01ffffff * lfe_fir_64[i]);

174 lfe_fir_64i[511 - i] = (int32_t)(0x01ffffff * lfe_fir_64[i]);

175 }

176

177 for (i = 0; i < 512; i++) {

178 band_interpolation[0][i] = (int32_t)(0x1000000000ULL * fir_32bands_perfect[i]);

179 band_interpolation[1][i] = (int32_t)(0x1000000000ULL * fir_32bands_nonperfect[i]);

180 }

181

182 for (i = 0; i < 9; i++) {

183 for (j = 0; j < AUBANDS; j++) {

184 for (k = 0; k < 256; k++) {

185 double freq = sample_rates[i] * (k + 0.5) / 512;

186

187 auf[i][j][k] = (int32_t)(10 * (hom(freq) + gammafilter(j, freq)));

188 }

189 }

190 }

191

192 for (i = 0; i < 256; i++) {

193 double add = 1 + pow(10, -0.01 * i);

194 cb_to_add[i] = (int32_t)(100 * log10(add));

195 }

196 for (j = 0; j < 8; j++) {

197 double accum = 0;

198 for (i = 0; i < 512; i++) {

199 double reconst = fir_32bands_perfect[i] * ((i & 64) ? (-1) : 1);

200 accum += reconst * cos(2 * M_PI * (i + 0.5 - 256) * (j + 0.5) / 512);

201 }

202 band_spectrum[0][j] = (int32_t)(200 * log10(accum));

203 }

204 for (j = 0; j < 8; j++) {

205 double accum = 0;

206 for (i = 0; i < 512; i++) {

207 double reconst = fir_32bands_nonperfect[i] * ((i & 64) ? (-1) : 1);

208 accum += reconst * cos(2 * M_PI * (i + 0.5 - 256) * (j + 0.5) / 512);

209 }

210 band_spectrum[1][j] = (int32_t)(200 * log10(accum));

211 }

212 }

213 return 0;

214 }

215

216 static inline int32_t cos_t(int x)

217 {

218 return cos_table[x & 2047];

219 }

220

221 static inline int32_t sin_t(int x)

222 {

223 return cos_t(x - 512);

224 }

225

226 static inline int32_t half32(int32_t a)

227 {

228 return (a + 1) >> 1;

229 }

230

231 static inline int32_t mul32(int32_t a, int32_t b)

232 {

233 int64_t r = (int64_t)a * b + 0x80000000ULL;

234 return r >> 32;

235 }

236

237 static void subband_transform(DCAContext *c, const int32_t *input)

238 {

239 int ch, subs, i, k, j;

240

241 for (ch = 0; ch < c->fullband_channels; ch++) {

242 /* History is copied because it is also needed for PSY */

243 int32_t hist[512];

244 int hist_start = 0;

245

246 for (i = 0; i < 512; i++)

247 hist[i] = c->history[i][ch];

248

249 for (subs = 0; subs < SUBBAND_SAMPLES; subs++) {

250 int32_t accum[64];

251 int32_t resp;

252 int band;

253

254 /* Calculate the convolutions at once */

255 for (i = 0; i < 64; i++)

256 accum[i] = 0;

257

258 for (k = 0, i = hist_start, j = 0;

259 i < 512; k = (k + 1) & 63, i++, j++)

260 accum[k] += mul32(hist[i], c->band_interpolation[j]);

261 for (i = 0; i < hist_start; k = (k + 1) & 63, i++, j++)

262 accum[k] += mul32(hist[i], c->band_interpolation[j]);

263

264 for (k = 16; k < 32; k++)

265 accum[k] = accum[k] - accum[31 - k];

266 for (k = 32; k < 48; k++)

267 accum[k] = accum[k] + accum[95 - k];

268

269 for (band = 0; band < 32; band++) {

270 resp = 0;

271 for (i = 16; i < 48; i++) {

272 int s = (2 * band + 1) * (2 * (i + 16) + 1);

273 resp += mul32(accum[i], cos_t(s << 3)) >> 3;

274 }

275

276 c->subband[subs][band][ch] = ((band + 1) & 2) ? -resp : resp;

277 }

278

279 /* Copy in 32 new samples from input */

280 for (i = 0; i < 32; i++)

281 hist[i + hist_start] = input[(subs * 32 + i) * c->channels + ch];

282 hist_start = (hist_start + 32) & 511;

283 }

284 }

285 }

286

287 static void lfe_downsample(DCAContext *c, const int32_t *input)

288 {

289 /* FIXME: make 128x LFE downsampling possible */

290 int i, j, lfes;

291 int32_t hist[512];

292 int32_t accum;

293 int hist_start = 0;

294

295 for (i = 0; i < 512; i++)

296 hist[i] = c->history[i][c->channels - 1];

297

298 for (lfes = 0; lfes < DCA_LFE_SAMPLES; lfes++) {

299 /* Calculate the convolution */

300 accum = 0;

301

302 for (i = hist_start, j = 0; i < 512; i++, j++)

303 accum += mul32(hist[i], lfe_fir_64i[j]);

304 for (i = 0; i < hist_start; i++, j++)

305 accum += mul32(hist[i], lfe_fir_64i[j]);

306

307 c->downsampled_lfe[lfes] = accum;

308

309 /* Copy in 64 new samples from input */

310 for (i = 0; i < 64; i++)

311 hist[i + hist_start] = input[(lfes * 64 + i) * c->channels + c->channels - 1];

312

313 hist_start = (hist_start + 64) & 511;

314 }

315 }

316

317 typedef struct {

318 int32_t re;

319 int32_t im;

320 } cplx32;

321

322 static void fft(const int32_t in[2 * 256], cplx32 out[256])

323 {

324 cplx32 buf[256], rin[256], rout[256];

325 int i, j, k, l;

326

327 /* do two transforms in parallel */

328 for (i = 0; i < 256; i++) {

329 /* Apply the Hann window */

330 rin[i].re = mul32(in[2 * i], 0x3fffffff - (cos_t(8 * i + 2) >> 1));

331 rin[i].im = mul32(in[2 * i + 1], 0x3fffffff - (cos_t(8 * i + 6) >> 1));

332 }

333 /* pre-rotation */

334 for (i = 0; i < 256; i++) {

335 buf[i].re = mul32(cos_t(4 * i + 2), rin[i].re)

336 - mul32(sin_t(4 * i + 2), rin[i].im);

337 buf[i].im = mul32(cos_t(4 * i + 2), rin[i].im)

338 + mul32(sin_t(4 * i + 2), rin[i].re);

339 }

340

341 for (j = 256, l = 1; j != 1; j >>= 1, l <<= 1) {

342 for (k = 0; k < 256; k += j) {

343 for (i = k; i < k + j / 2; i++) {

344 cplx32 sum, diff;

345 int t = 8 * l * i;

346

347 sum.re = buf[i].re + buf[i + j / 2].re;

348 sum.im = buf[i].im + buf[i + j / 2].im;

349

350 diff.re = buf[i].re - buf[i + j / 2].re;

351 diff.im = buf[i].im - buf[i + j / 2].im;

352

353 buf[i].re = half32(sum.re);

354 buf[i].im = half32(sum.im);

355

356 buf[i + j / 2].re = mul32(diff.re, cos_t(t))

357 - mul32(diff.im, sin_t(t));

358 buf[i + j / 2].im = mul32(diff.im, cos_t(t))

359 + mul32(diff.re, sin_t(t));

360 }

361 }

362 }

363 /* post-rotation */

364 for (i = 0; i < 256; i++) {

365 int b = ff_reverse[i];

366 rout[i].re = mul32(buf[b].re, cos_t(4 * i))

367 - mul32(buf[b].im, sin_t(4 * i));

368 rout[i].im = mul32(buf[b].im, cos_t(4 * i))

369 + mul32(buf[b].re, sin_t(4 * i));

370 }

371 for (i = 0; i < 256; i++) {

372 /* separate the results of the two transforms */

373 cplx32 o1, o2;

374

375 o1.re = rout[i].re - rout[255 - i].re;

376 o1.im = rout[i].im + rout[255 - i].im;

377

378 o2.re = rout[i].im - rout[255 - i].im;

379 o2.im = -rout[i].re - rout[255 - i].re;

380

381 /* combine them into one long transform */

382 out[i].re = mul32( o1.re + o2.re, cos_t(2 * i + 1))

383 + mul32( o1.im - o2.im, sin_t(2 * i + 1));

384 out[i].im = mul32( o1.im + o2.im, cos_t(2 * i + 1))

385 + mul32(-o1.re + o2.re, sin_t(2 * i + 1));

386 }

387 }

388

389 static int32_t get_cb(int32_t in)

390 {

391 int i, res;

392

393 res = 0;

394 if (in < 0)

395 in = -in;

396 for (i = 1024; i > 0; i >>= 1) {

397 if (cb_to_level[i + res] >= in)

398 res += i;

399 }

400 return -res;

401 }

402

403 static int32_t add_cb(int32_t a, int32_t b)

404 {

405 if (a < b)

406 FFSWAP(int32_t, a, b);

407

408 if (a - b >= 256)

409 return a;

410 return a + cb_to_add[a - b];

411 }

412

413 static void adjust_jnd(int samplerate_index,

414 const int32_t in[512], int32_t out_cb[256])

415 {

416 int32_t power[256];

417 cplx32 out[256];

418 int32_t out_cb_unnorm[256];

419 int32_t denom;

420 const int32_t ca_cb = -1114;

421 const int32_t cs_cb = 928;

422 int i, j;

423

424 fft(in, out);

425

426 for (j = 0; j < 256; j++) {

427 power[j] = add_cb(get_cb(out[j].re), get_cb(out[j].im));

428 out_cb_unnorm[j] = -2047; /* and can only grow */

429 }

430

431 for (i = 0; i < AUBANDS; i++) {

432 denom = ca_cb; /* and can only grow */

433 for (j = 0; j < 256; j++)

434 denom = add_cb(denom, power[j] + auf[samplerate_index][i][j]);

435 for (j = 0; j < 256; j++)

436 out_cb_unnorm[j] = add_cb(out_cb_unnorm[j],

437 -denom + auf[samplerate_index][i][j]);

438 }

439

440 for (j = 0; j < 256; j++)

441 out_cb[j] = add_cb(out_cb[j], -out_cb_unnorm[j] - ca_cb - cs_cb);

442 }

443

444 typedef void (*walk_band_t)(DCAContext *c, int band1, int band2, int f,

445 int32_t spectrum1, int32_t spectrum2, int channel,

446 int32_t * arg);

447

448 static void walk_band_low(DCAContext *c, int band, int channel,

449 walk_band_t walk, int32_t *arg)

450 {

451 int f;

452

453 if (band == 0) {

454 for (f = 0; f < 4; f++)

455 walk(c, 0, 0, f, 0, -2047, channel, arg);

456 } else {

457 for (f = 0; f < 8; f++)

458 walk(c, band, band - 1, 8 * band - 4 + f,

459 c->band_spectrum[7 - f], c->band_spectrum[f], channel, arg);

460 }

461 }

462

463 static void walk_band_high(DCAContext *c, int band, int channel,

464 walk_band_t walk, int32_t *arg)

465 {

466 int f;

467

468 if (band == 31) {

469 for (f = 0; f < 4; f++)

470 walk(c, 31, 31, 256 - 4 + f, 0, -2047, channel, arg);

471 } else {

472 for (f = 0; f < 8; f++)

473 walk(c, band, band + 1, 8 * band + 4 + f,

474 c->band_spectrum[f], c->band_spectrum[7 - f], channel, arg);

475 }

476 }

477

478 static void update_band_masking(DCAContext *c, int band1, int band2,

479 int f, int32_t spectrum1, int32_t spectrum2,

480 int channel, int32_t * arg)

481 {

482 int32_t value = c->eff_masking_curve_cb[f] - spectrum1;

483

484 if (value < c->band_masking_cb[band1])

485 c->band_masking_cb[band1] = value;

486 }

487

488 static void calc_masking(DCAContext *c, const int32_t *input)

489 {

490 int i, k, band, ch, ssf;

491 int32_t data[512];

492

493 for (i = 0; i < 256; i++)

494 for (ssf = 0; ssf < SUBSUBFRAMES; ssf++)

495 c->masking_curve_cb[ssf][i] = -2047;

496

497 for (ssf = 0; ssf < SUBSUBFRAMES; ssf++)

498 for (ch = 0; ch < c->fullband_channels; ch++) {

499 for (i = 0, k = 128 + 256 * ssf; k < 512; i++, k++)

500 data[i] = c->history[k][ch];

501 for (k -= 512; i < 512; i++, k++)

502 data[i] = input[k * c->channels + ch];

503 adjust_jnd(c->samplerate_index, data, c->masking_curve_cb[ssf]);

504 }

505 for (i = 0; i < 256; i++) {

506 int32_t m = 2048;

507

508 for (ssf = 0; ssf < SUBSUBFRAMES; ssf++)

509 if (c->masking_curve_cb[ssf][i] < m)

510 m = c->masking_curve_cb[ssf][i];

511 c->eff_masking_curve_cb[i] = m;

512 }

513

514 for (band = 0; band < 32; band++) {

515 c->band_masking_cb[band] = 2048;

516 walk_band_low(c, band, 0, update_band_masking, NULL);

517 walk_band_high(c, band, 0, update_band_masking, NULL);

518 }

519 }

520

521 static void find_peaks(DCAContext *c)

522 {

523 int band, ch;

524

525 for (band = 0; band < 32; band++)

526 for (ch = 0; ch < c->fullband_channels; ch++) {

527 int sample;

528 int32_t m = 0;

529

530 for (sample = 0; sample < SUBBAND_SAMPLES; sample++) {

531 int32_t s = abs(c->subband[sample][band][ch]);

532 if (m < s)

533 m = s;

534 }

535 c->peak_cb[band][ch] = get_cb(m);

536 }

537

538 if (c->lfe_channel) {

539 int sample;

540 int32_t m = 0;

541

542 for (sample = 0; sample < DCA_LFE_SAMPLES; sample++)

543 if (m < abs(c->downsampled_lfe[sample]))

544 m = abs(c->downsampled_lfe[sample]);

545 c->lfe_peak_cb = get_cb(m);

546 }

547 }

548

549 static const int snr_fudge = 128;

550 #define USED_1ABITS 1

551 #define USED_NABITS 2

552 #define USED_26ABITS 4

553

554 static int init_quantization_noise(DCAContext *c, int noise)

555 {

556 int ch, band, ret = 0;

557

558 c->consumed_bits = 132 + 493 * c->fullband_channels;

559 if (c->lfe_channel)

560 c->consumed_bits += 72;

561

562 /* attempt to guess the bit distribution based on the prevoius frame */

563 for (ch = 0; ch < c->fullband_channels; ch++) {

564 for (band = 0; band < 32; band++) {

565 int snr_cb = c->peak_cb[band][ch] - c->band_masking_cb[band] - noise;

566

567 if (snr_cb >= 1312) {

568 c->abits[band][ch] = 26;

569 ret |= USED_26ABITS;

570 } else if (snr_cb >= 222) {

571 c->abits[band][ch] = 8 + mul32(snr_cb - 222, 69000000);

572 ret |= USED_NABITS;

573 } else if (snr_cb >= 0) {

574 c->abits[band][ch] = 2 + mul32(snr_cb, 106000000);

575 ret |= USED_NABITS;

576 } else {

577 c->abits[band][ch] = 1;

578 ret |= USED_1ABITS;

579 }

580 }

581 }

582

583 for (band = 0; band < 32; band++)

584 for (ch = 0; ch < c->fullband_channels; ch++) {

585 c->consumed_bits += bit_consumption[c->abits[band][ch]];

586 }

587

588 return ret;

589 }

590

591 static void assign_bits(DCAContext *c)

592 {

593 /* Find the bounds where the binary search should work */

594 int low, high, down;

595 int used_abits = 0;

596

597 init_quantization_noise(c, c->worst_quantization_noise);

598 low = high = c->worst_quantization_noise;

599 if (c->consumed_bits > c->frame_bits) {

600 while (c->consumed_bits > c->frame_bits) {

601 av_assert0(used_abits != USED_1ABITS);

602 low = high;

603 high += snr_fudge;

604 used_abits = init_quantization_noise(c, high);

605 }

606 } else {

607 while (c->consumed_bits <= c->frame_bits) {

608 high = low;

609 if (used_abits == USED_26ABITS)

610 goto out; /* The requested bitrate is too high, pad with zeros */

611 low -= snr_fudge;

612 used_abits = init_quantization_noise(c, low);

613 }

614 }

615

616 /* Now do a binary search between low and high to see what fits */

617 for (down = snr_fudge >> 1; down; down >>= 1) {

618 init_quantization_noise(c, high - down);

619 if (c->consumed_bits <= c->frame_bits)

620 high -= down;

621 }

622 init_quantization_noise(c, high);

623 out:

624 c->worst_quantization_noise = high;

625 if (high > c->worst_noise_ever)

626 c->worst_noise_ever = high;

627 }

628

629 static void shift_history(DCAContext *c, const int32_t *input)

630 {

631 int k, ch;

632

633 for (k = 0; k < 512; k++)

634 for (ch = 0; ch < c->channels; ch++)

635 c->history[k][ch] = input[k * c->channels + ch];

636 }

637

638 static int32_t quantize_value(int32_t value, softfloat quant)

639 {

640 int32_t offset = 1 << (quant.e - 1);

641

642 value = mul32(value, quant.m) + offset;

643 value = value >> quant.e;

644 return value;

645 }

646

647 static int calc_one_scale(int32_t peak_cb, int abits, softfloat *quant)

648 {

649 int32_t peak;

650 int our_nscale, try_remove;

651 softfloat our_quant;

652

653 av_assert0(peak_cb <= 0);

654 av_assert0(peak_cb >= -2047);

655

656 our_nscale = 127;

657 peak = cb_to_level[-peak_cb];

658

659 for (try_remove = 64; try_remove > 0; try_remove >>= 1) {

660 if (scalefactor_inv[our_nscale - try_remove].e + stepsize_inv[abits].e <= 17)

661 continue;

662 our_quant.m = mul32(scalefactor_inv[our_nscale - try_remove].m, stepsize_inv[abits].m);

663 our_quant.e = scalefactor_inv[our_nscale - try_remove].e + stepsize_inv[abits].e - 17;

664 if ((quant_levels[abits] - 1) / 2 < quantize_value(peak, our_quant))

665 continue;

666 our_nscale -= try_remove;

667 }

668

669 if (our_nscale >= 125)

670 our_nscale = 124;

671

672 quant->m = mul32(scalefactor_inv[our_nscale].m, stepsize_inv[abits].m);

673 quant->e = scalefactor_inv[our_nscale].e + stepsize_inv[abits].e - 17;

674 av_assert0((quant_levels[abits] - 1) / 2 >= quantize_value(peak, *quant));

675

676 return our_nscale;

677 }

678

679 static void calc_scales(DCAContext *c)

680 {

681 int band, ch;

682

683 for (band = 0; band < 32; band++)

684 for (ch = 0; ch < c->fullband_channels; ch++)

685 c->scale_factor[band][ch] = calc_one_scale(c->peak_cb[band][ch],

686 c->abits[band][ch],

687 &c->quant[band][ch]);

688

689 if (c->lfe_channel)

690 c->lfe_scale_factor = calc_one_scale(c->lfe_peak_cb, 11, &c->lfe_quant);

691 }

692

693 static void quantize_all(DCAContext *c)

694 {

695 int sample, band, ch;

696

697 for (sample = 0; sample < SUBBAND_SAMPLES; sample++)

698 for (band = 0; band < 32; band++)

699 for (ch = 0; ch < c->fullband_channels; ch++)

700 c->quantized[sample][band][ch] = quantize_value(c->subband[sample][band][ch], c->quant[band][ch]);

701 }

702

703 static void put_frame_header(DCAContext *c)

704 {

705 /* SYNC */

706 put_bits(&c->pb, 16, 0x7ffe);

707 put_bits(&c->pb, 16, 0x8001);

708

709 /* Frame type: normal */

710 put_bits(&c->pb, 1, 1);

711

712 /* Deficit sample count: none */

713 put_bits(&c->pb, 5, 31);

714

715 /* CRC is not present */

716 put_bits(&c->pb, 1, 0);

717

718 /* Number of PCM sample blocks */

719 put_bits(&c->pb, 7, SUBBAND_SAMPLES - 1);

720

721 /* Primary frame byte size */

722 put_bits(&c->pb, 14, c->frame_size - 1);

723

724 /* Audio channel arrangement */

725 put_bits(&c->pb, 6, c->channel_config);

726

727 /* Core audio sampling frequency */

728 put_bits(&c->pb, 4, bitstream_sfreq[c->samplerate_index]);

729

730 /* Transmission bit rate */

731 put_bits(&c->pb, 5, c->bitrate_index);

732

733 /* Embedded down mix: disabled */

734 put_bits(&c->pb, 1, 0);

735

736 /* Embedded dynamic range flag: not present */

737 put_bits(&c->pb, 1, 0);

738

739 /* Embedded time stamp flag: not present */

740 put_bits(&c->pb, 1, 0);

741

742 /* Auxiliary data flag: not present */

743 put_bits(&c->pb, 1, 0);

744

745 /* HDCD source: no */

746 put_bits(&c->pb, 1, 0);

747

748 /* Extension audio ID: N/A */

749 put_bits(&c->pb, 3, 0);

750

751 /* Extended audio data: not present */

752 put_bits(&c->pb, 1, 0);

753

754 /* Audio sync word insertion flag: after each sub-frame */

755 put_bits(&c->pb, 1, 0);

756

757 /* Low frequency effects flag: not present or 64x subsampling */

758 put_bits(&c->pb, 2, c->lfe_channel ? 2 : 0);

759

760 /* Predictor history switch flag: on */

761 put_bits(&c->pb, 1, 1);

762

763 /* No CRC */

764 /* Multirate interpolator switch: non-perfect reconstruction */

765 put_bits(&c->pb, 1, 0);

766

767 /* Encoder software revision: 7 */

768 put_bits(&c->pb, 4, 7);

769

770 /* Copy history: 0 */

771 put_bits(&c->pb, 2, 0);

772

773 /* Source PCM resolution: 16 bits, not DTS ES */

774 put_bits(&c->pb, 3, 0);

775

776 /* Front sum/difference coding: no */

777 put_bits(&c->pb, 1, 0);

778

779 /* Surrounds sum/difference coding: no */

780 put_bits(&c->pb, 1, 0);

781

782 /* Dialog normalization: 0 dB */

783 put_bits(&c->pb, 4, 0);

784 }

785

786 static void put_primary_audio_header(DCAContext *c)

787 {

788 static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 };

789 static const int thr[11] = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 };

790

791 int ch, i;

792 /* Number of subframes */

793 put_bits(&c->pb, 4, SUBFRAMES - 1);

794

795 /* Number of primary audio channels */

796 put_bits(&c->pb, 3, c->fullband_channels - 1);

797

798 /* Subband activity count */

799 for (ch = 0; ch < c->fullband_channels; ch++)

800 put_bits(&c->pb, 5, DCA_SUBBANDS - 2);

801

802 /* High frequency VQ start subband */

803 for (ch = 0; ch < c->fullband_channels; ch++)

804 put_bits(&c->pb, 5, DCA_SUBBANDS - 1);

805

806 /* Joint intensity coding index: 0, 0 */

807 for (ch = 0; ch < c->fullband_channels; ch++)

808 put_bits(&c->pb, 3, 0);

809

810 /* Transient mode codebook: A4, A4 (arbitrary) */

811 for (ch = 0; ch < c->fullband_channels; ch++)

812 put_bits(&c->pb, 2, 0);

813

814 /* Scale factor code book: 7 bit linear, 7-bit sqrt table (for each channel) */

815 for (ch = 0; ch < c->fullband_channels; ch++)

816 put_bits(&c->pb, 3, 6);

817

818 /* Bit allocation quantizer select: linear 5-bit */

819 for (ch = 0; ch < c->fullband_channels; ch++)

820 put_bits(&c->pb, 3, 6);

821

822 /* Quantization index codebook select: dummy data

823 to avoid transmission of scale factor adjustment */

824 for (i = 1; i < 11; i++)

825 for (ch = 0; ch < c->fullband_channels; ch++)

826 put_bits(&c->pb, bitlen[i], thr[i]);

827

828 /* Scale factor adjustment index: not transmitted */

829 /* Audio header CRC check word: not transmitted */

830 }

831

832 static void put_subframe_samples(DCAContext *c, int ss, int band, int ch)

833 {

834 if (c->abits[band][ch] <= 7) {

835 int sum, i, j;

836 for (i = 0; i < 8; i += 4) {

837 sum = 0;

838 for (j = 3; j >= 0; j--) {

839 sum *= quant_levels[c->abits[band][ch]];

840 sum += c->quantized[ss * 8 + i + j][band][ch];

841 sum += (quant_levels[c->abits[band][ch]] - 1) / 2;

842 }

843 put_bits(&c->pb, bit_consumption[c->abits[band][ch]] / 4, sum);

844 }

845 } else {

846 int i;

847 for (i = 0; i < 8; i++) {

848 int bits = bit_consumption[c->abits[band][ch]] / 16;

849 int32_t mask = (1 << bits) - 1;

850 put_bits(&c->pb, bits, c->quantized[ss * 8 + i][band][ch] & mask);

851 }

852 }

853 }

854

855 static void put_subframe(DCAContext *c, int subframe)

856 {

857 int i, band, ss, ch;

858

859 /* Subsubframes count */

860 put_bits(&c->pb, 2, SUBSUBFRAMES -1);

861

862 /* Partial subsubframe sample count: dummy */

863 put_bits(&c->pb, 3, 0);

864

865 /* Prediction mode: no ADPCM, in each channel and subband */

866 for (ch = 0; ch < c->fullband_channels; ch++)

867 for (band = 0; band < DCA_SUBBANDS; band++)

868 put_bits(&c->pb, 1, 0);

869

870 /* Prediction VQ addres: not transmitted */

871 /* Bit allocation index */

872 for (ch = 0; ch < c->fullband_channels; ch++)

873 for (band = 0; band < DCA_SUBBANDS; band++)

874 put_bits(&c->pb, 5, c->abits[band][ch]);

875

876 if (SUBSUBFRAMES > 1) {

877 /* Transition mode: none for each channel and subband */

878 for (ch = 0; ch < c->fullband_channels; ch++)

879 for (band = 0; band < DCA_SUBBANDS; band++)

880 put_bits(&c->pb, 1, 0); /* codebook A4 */

881 }

882

883 /* Scale factors */

884 for (ch = 0; ch < c->fullband_channels; ch++)

885 for (band = 0; band < DCA_SUBBANDS; band++)

886 put_bits(&c->pb, 7, c->scale_factor[band][ch]);

887

888 /* Joint subband scale factor codebook select: not transmitted */

889 /* Scale factors for joint subband coding: not transmitted */

890 /* Stereo down-mix coefficients: not transmitted */

891 /* Dynamic range coefficient: not transmitted */

892 /* Stde information CRC check word: not transmitted */

893 /* VQ encoded high frequency subbands: not transmitted */

894

895 /* LFE data: 8 samples and scalefactor */

896 if (c->lfe_channel) {

897 for (i = 0; i < DCA_LFE_SAMPLES; i++)

898 put_bits(&c->pb, 8, quantize_value(c->downsampled_lfe[i], c->lfe_quant) & 0xff);

899 put_bits(&c->pb, 8, c->lfe_scale_factor);

900 }

901

902 /* Audio data (subsubframes) */

903 for (ss = 0; ss < SUBSUBFRAMES ; ss++)

904 for (ch = 0; ch < c->fullband_channels; ch++)

905 for (band = 0; band < DCA_SUBBANDS; band++)

906 put_subframe_samples(c, ss, band, ch);

907

908 /* DSYNC */

909 put_bits(&c->pb, 16, 0xffff);

910 }

911

912 static int encode_frame(AVCodecContext *avctx, AVPacket *avpkt,

913 const AVFrame *frame, int *got_packet_ptr)

914 {

915 DCAContext *c = avctx->priv_data;

916 const int32_t *samples;

917 int ret, i;

918

919 if ((ret = ff_alloc_packet2(avctx, avpkt, c->frame_size )) < 0)

920 return ret;

921

922 samples = (const int32_t *)frame->data[0];

923

924 subband_transform(c, samples);

925 if (c->lfe_channel)

926 lfe_downsample(c, samples);

927

928 calc_masking(c, samples);

929 find_peaks(c);

930 assign_bits(c);

931 calc_scales(c);

932 quantize_all(c);

933 shift_history(c, samples);

934

935 init_put_bits(&c->pb, avpkt->data, avpkt->size);

936 put_frame_header(c);

937 put_primary_audio_header(c);

938 for (i = 0; i < SUBFRAMES; i++)

939 put_subframe(c, i);

940

941 flush_put_bits(&c->pb);

942

943 avpkt->pts = frame->pts;

944 avpkt->duration = ff_samples_to_time_base(avctx, frame->nb_samples);

945 avpkt->size = c->frame_size + 1;

946 *got_packet_ptr = 1;

947 return 0;

948 }

949

950 static const AVCodecDefault defaults[] = {

951 { "b", "1411200" },

952 { NULL },

953 };

954

955 AVCodec ff_dca_encoder = {

956 .name = "dca",

957 .type = AVMEDIA_TYPE_AUDIO,

958 .id = AV_CODEC_ID_DTS,

959 .priv_data_size = sizeof(DCAContext),

960 .init = encode_init,

961 .encode2 = encode_frame,

962 .capabilities = CODEC_CAP_EXPERIMENTAL,

963 .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S32,

964 AV_SAMPLE_FMT_NONE },

965 .long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),

966 .supported_samplerates = sample_rates,

967 .channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_MONO,

968 AV_CH_LAYOUT_STEREO,

969 AV_CH_LAYOUT_2_2,

970 AV_CH_LAYOUT_5POINT0,

971 AV_CH_LAYOUT_5POINT1,

972 0 },

973 .defaults = defaults,

974 };

Generated on Wed Jul 10 2013 23:47:56 for FFmpeg by doxygen 1.8.2