FFmpeg: libavcodec/amrwbdec.c Source File

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amrwbdec.c

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1 /*

2 * AMR wideband decoder

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 * AMR wideband decoder

25 */

27 #include "libavutil/channel_layout.h"

28 #include "libavutil/common.h"

29 #include "libavutil/float_dsp.h"

30 #include "libavutil/lfg.h"

32 #include "avcodec.h"

33 #include "lsp.h"

34 #include "celp_filters.h"

35 #include "celp_math.h"

36 #include "acelp_filters.h"

37 #include "acelp_vectors.h"

38 #include "acelp_pitch_delay.h"

39 #include "internal.h"

41 #define AMR_USE_16BIT_TABLES

42 #include "amr.h"

44 #include "amrwbdata.h"

45 #include "mips/amrwbdec_mips.h"

47 typedef struct {

48 AMRWBFrame frame; ///< AMRWB parameters decoded from bitstream

49 enum Mode fr_cur_mode; ///< mode index of current frame

50 uint8_t fr_quality; ///< frame quality index (FQI)

51 float isf_cur[LP_ORDER]; ///< working ISF vector from current frame

52 float isf_q_past[LP_ORDER]; ///< quantized ISF vector of the previous frame

53 float isf_past_final[LP_ORDER]; ///< final processed ISF vector of the previous frame

54 double isp[4][LP_ORDER]; ///< ISP vectors from current frame

55 double isp_sub4_past[LP_ORDER]; ///< ISP vector for the 4th subframe of the previous frame

57 float lp_coef[4][LP_ORDER]; ///< Linear Prediction Coefficients from ISP vector

59 uint8_t base_pitch_lag; ///< integer part of pitch lag for the next relative subframe

60 uint8_t pitch_lag_int; ///< integer part of pitch lag of the previous subframe

62 float excitation_buf[AMRWB_P_DELAY_MAX + LP_ORDER + 2 + AMRWB_SFR_SIZE]; ///< current excitation and all necessary excitation history

63 float *excitation; ///< points to current excitation in excitation_buf[]

65 float pitch_vector[AMRWB_SFR_SIZE]; ///< adaptive codebook (pitch) vector for current subframe

66 float fixed_vector[AMRWB_SFR_SIZE]; ///< algebraic codebook (fixed) vector for current subframe

68 float prediction_error[4]; ///< quantified prediction errors {20log10(^gamma_gc)} for previous four subframes

69 float pitch_gain[6]; ///< quantified pitch gains for the current and previous five subframes

70 float fixed_gain[2]; ///< quantified fixed gains for the current and previous subframes

72 float tilt_coef; ///< {beta_1} related to the voicing of the previous subframe

74 float prev_sparse_fixed_gain; ///< previous fixed gain; used by anti-sparseness to determine "onset"

75 uint8_t prev_ir_filter_nr; ///< previous impulse response filter "impNr": 0 - strong, 1 - medium, 2 - none

76 float prev_tr_gain; ///< previous initial gain used by noise enhancer for threshold

78 float samples_az[LP_ORDER + AMRWB_SFR_SIZE]; ///< low-band samples and memory from synthesis at 12.8kHz

79 float samples_up[UPS_MEM_SIZE + AMRWB_SFR_SIZE]; ///< low-band samples and memory processed for upsampling

80 float samples_hb[LP_ORDER_16k + AMRWB_SFR_SIZE_16k]; ///< high-band samples and memory from synthesis at 16kHz

82 float hpf_31_mem[2], hpf_400_mem[2]; ///< previous values in the high pass filters

83 float demph_mem[1]; ///< previous value in the de-emphasis filter

84 float bpf_6_7_mem[HB_FIR_SIZE]; ///< previous values in the high-band band pass filter

85 float lpf_7_mem[HB_FIR_SIZE]; ///< previous values in the high-band low pass filter

87 AVLFG prng; ///< random number generator for white noise excitation

88 uint8_t first_frame; ///< flag active during decoding of the first frame

89 ACELPFContext acelpf_ctx; ///< context for filters for ACELP-based codecs

90 ACELPVContext acelpv_ctx; ///< context for vector operations for ACELP-based codecs

91 CELPFContext celpf_ctx; ///< context for filters for CELP-based codecs

92 CELPMContext celpm_ctx; ///< context for fixed point math operations

94 } AMRWBContext;

96 static av_cold int amrwb_decode_init(AVCodecContext *avctx)

97 {

98 AMRWBContext *ctx = avctx->priv_data;

99 int i;

100

101 if (avctx->channels > 1) {

102 avpriv_report_missing_feature(avctx, "multi-channel AMR");

103 return AVERROR_PATCHWELCOME;

104 }

105

106 avctx->channels = 1;

107 avctx->channel_layout = AV_CH_LAYOUT_MONO;

108 if (!avctx->sample_rate)

109 avctx->sample_rate = 16000;

110 avctx->sample_fmt = AV_SAMPLE_FMT_FLT;

111

112 av_lfg_init(&ctx->prng, 1);

113

114 ctx->excitation = &ctx->excitation_buf[AMRWB_P_DELAY_MAX + LP_ORDER + 1];

115 ctx->first_frame = 1;

116

117 for (i = 0; i < LP_ORDER; i++)

118 ctx->isf_past_final[i] = isf_init[i] * (1.0f / (1 << 15));

119

120 for (i = 0; i < 4; i++)

121 ctx->prediction_error[i] = MIN_ENERGY;

122

123 ff_acelp_filter_init(&ctx->acelpf_ctx);

124 ff_acelp_vectors_init(&ctx->acelpv_ctx);

125 ff_celp_filter_init(&ctx->celpf_ctx);

126 ff_celp_math_init(&ctx->celpm_ctx);

127

128 return 0;

129 }

130

131 /**

132 * Decode the frame header in the "MIME/storage" format. This format

133 * is simpler and does not carry the auxiliary frame information.

134 *

135 * @param[in] ctx The Context

136 * @param[in] buf Pointer to the input buffer

137 *

138 * @return The decoded header length in bytes

139 */

140 static int decode_mime_header(AMRWBContext *ctx, const uint8_t *buf)

141 {

142 /* Decode frame header (1st octet) */

143 ctx->fr_cur_mode = buf[0] >> 3 & 0x0F;

144 ctx->fr_quality = (buf[0] & 0x4) == 0x4;

145

146 return 1;

147 }

148

149 /**

150 * Decode quantized ISF vectors using 36-bit indexes (6K60 mode only).

151 *

152 * @param[in] ind Array of 5 indexes

153 * @param[out] isf_q Buffer for isf_q[LP_ORDER]

154 *

155 */

156 static void decode_isf_indices_36b(uint16_t *ind, float *isf_q)

157 {

158 int i;

159

160 for (i = 0; i < 9; i++)

161 isf_q[i] = dico1_isf[ind[0]][i] * (1.0f / (1 << 15));

162

163 for (i = 0; i < 7; i++)

164 isf_q[i + 9] = dico2_isf[ind[1]][i] * (1.0f / (1 << 15));

165

166 for (i = 0; i < 5; i++)

167 isf_q[i] += dico21_isf_36b[ind[2]][i] * (1.0f / (1 << 15));

168

169 for (i = 0; i < 4; i++)

170 isf_q[i + 5] += dico22_isf_36b[ind[3]][i] * (1.0f / (1 << 15));

171

172 for (i = 0; i < 7; i++)

173 isf_q[i + 9] += dico23_isf_36b[ind[4]][i] * (1.0f / (1 << 15));

174 }

175

176 /**

177 * Decode quantized ISF vectors using 46-bit indexes (except 6K60 mode).

178 *

179 * @param[in] ind Array of 7 indexes

180 * @param[out] isf_q Buffer for isf_q[LP_ORDER]

181 *

182 */

183 static void decode_isf_indices_46b(uint16_t *ind, float *isf_q)

184 {

185 int i;

186

187 for (i = 0; i < 9; i++)

188 isf_q[i] = dico1_isf[ind[0]][i] * (1.0f / (1 << 15));

189

190 for (i = 0; i < 7; i++)

191 isf_q[i + 9] = dico2_isf[ind[1]][i] * (1.0f / (1 << 15));

192

193 for (i = 0; i < 3; i++)

194 isf_q[i] += dico21_isf[ind[2]][i] * (1.0f / (1 << 15));

195

196 for (i = 0; i < 3; i++)

197 isf_q[i + 3] += dico22_isf[ind[3]][i] * (1.0f / (1 << 15));

198

199 for (i = 0; i < 3; i++)

200 isf_q[i + 6] += dico23_isf[ind[4]][i] * (1.0f / (1 << 15));

201

202 for (i = 0; i < 3; i++)

203 isf_q[i + 9] += dico24_isf[ind[5]][i] * (1.0f / (1 << 15));

204

205 for (i = 0; i < 4; i++)

206 isf_q[i + 12] += dico25_isf[ind[6]][i] * (1.0f / (1 << 15));

207 }

208

209 /**

210 * Apply mean and past ISF values using the prediction factor.

211 * Updates past ISF vector.

212 *

213 * @param[in,out] isf_q Current quantized ISF

214 * @param[in,out] isf_past Past quantized ISF

215 *

216 */

217 static void isf_add_mean_and_past(float *isf_q, float *isf_past)

218 {

219 int i;

220 float tmp;

221

222 for (i = 0; i < LP_ORDER; i++) {

223 tmp = isf_q[i];

224 isf_q[i] += isf_mean[i] * (1.0f / (1 << 15));

225 isf_q[i] += PRED_FACTOR * isf_past[i];

226 isf_past[i] = tmp;

227 }

228 }

229

230 /**

231 * Interpolate the fourth ISP vector from current and past frames

232 * to obtain an ISP vector for each subframe.

233 *

234 * @param[in,out] isp_q ISPs for each subframe

235 * @param[in] isp4_past Past ISP for subframe 4

236 */

237 static void interpolate_isp(double isp_q[4][LP_ORDER], const double *isp4_past)

238 {

239 int i, k;

240

241 for (k = 0; k < 3; k++) {

242 float c = isfp_inter[k];

243 for (i = 0; i < LP_ORDER; i++)

244 isp_q[k][i] = (1.0 - c) * isp4_past[i] + c * isp_q[3][i];

245 }

246 }

247

248 /**

249 * Decode an adaptive codebook index into pitch lag (except 6k60, 8k85 modes).

250 * Calculate integer lag and fractional lag always using 1/4 resolution.

251 * In 1st and 3rd subframes the index is relative to last subframe integer lag.

252 *

253 * @param[out] lag_int Decoded integer pitch lag

254 * @param[out] lag_frac Decoded fractional pitch lag

255 * @param[in] pitch_index Adaptive codebook pitch index

256 * @param[in,out] base_lag_int Base integer lag used in relative subframes

257 * @param[in] subframe Current subframe index (0 to 3)

258 */

259 static void decode_pitch_lag_high(int *lag_int, int *lag_frac, int pitch_index,

260 uint8_t *base_lag_int, int subframe)

261 {

262 if (subframe == 0 || subframe == 2) {

263 if (pitch_index < 376) {

264 *lag_int = (pitch_index + 137) >> 2;

265 *lag_frac = pitch_index - (*lag_int << 2) + 136;

266 } else if (pitch_index < 440) {

267 *lag_int = (pitch_index + 257 - 376) >> 1;

268 *lag_frac = (pitch_index - (*lag_int << 1) + 256 - 376) << 1;

269 /* the actual resolution is 1/2 but expressed as 1/4 */

270 } else {

271 *lag_int = pitch_index - 280;

272 *lag_frac = 0;

273 }

274 /* minimum lag for next subframe */

275 *base_lag_int = av_clip(*lag_int - 8 - (*lag_frac < 0),

276 AMRWB_P_DELAY_MIN, AMRWB_P_DELAY_MAX - 15);

277 // XXX: the spec states clearly that *base_lag_int should be

278 // the nearest integer to *lag_int (minus 8), but the ref code

279 // actually always uses its floor, I'm following the latter

280 } else {

281 *lag_int = (pitch_index + 1) >> 2;

282 *lag_frac = pitch_index - (*lag_int << 2);

283 *lag_int += *base_lag_int;

284 }

285 }

286

287 /**

288 * Decode an adaptive codebook index into pitch lag for 8k85 and 6k60 modes.

289 * The description is analogous to decode_pitch_lag_high, but in 6k60 the

290 * relative index is used for all subframes except the first.

291 */

292 static void decode_pitch_lag_low(int *lag_int, int *lag_frac, int pitch_index,

293 uint8_t *base_lag_int, int subframe, enum Mode mode)

294 {

295 if (subframe == 0 || (subframe == 2 && mode != MODE_6k60)) {

296 if (pitch_index < 116) {

297 *lag_int = (pitch_index + 69) >> 1;

298 *lag_frac = (pitch_index - (*lag_int << 1) + 68) << 1;

299 } else {

300 *lag_int = pitch_index - 24;

301 *lag_frac = 0;

302 }

303 // XXX: same problem as before

304 *base_lag_int = av_clip(*lag_int - 8 - (*lag_frac < 0),

305 AMRWB_P_DELAY_MIN, AMRWB_P_DELAY_MAX - 15);

306 } else {

307 *lag_int = (pitch_index + 1) >> 1;

308 *lag_frac = (pitch_index - (*lag_int << 1)) << 1;

309 *lag_int += *base_lag_int;

310 }

311 }

312

313 /**

314 * Find the pitch vector by interpolating the past excitation at the

315 * pitch delay, which is obtained in this function.

316 *

317 * @param[in,out] ctx The context

318 * @param[in] amr_subframe Current subframe data

319 * @param[in] subframe Current subframe index (0 to 3)

320 */

321 static void decode_pitch_vector(AMRWBContext *ctx,

322 const AMRWBSubFrame *amr_subframe,

323 const int subframe)

324 {

325 int pitch_lag_int, pitch_lag_frac;

326 int i;

327 float *exc = ctx->excitation;

328 enum Mode mode = ctx->fr_cur_mode;

329

330 if (mode <= MODE_8k85) {

331 decode_pitch_lag_low(&pitch_lag_int, &pitch_lag_frac, amr_subframe->adap,

332 &ctx->base_pitch_lag, subframe, mode);

333 } else

334 decode_pitch_lag_high(&pitch_lag_int, &pitch_lag_frac, amr_subframe->adap,

335 &ctx->base_pitch_lag, subframe);

336

337 ctx->pitch_lag_int = pitch_lag_int;

338 pitch_lag_int += pitch_lag_frac > 0;

339

340 /* Calculate the pitch vector by interpolating the past excitation at the

341 pitch lag using a hamming windowed sinc function */

342 ctx->acelpf_ctx.acelp_interpolatef(exc,

343 exc + 1 - pitch_lag_int,

344 ac_inter, 4,

345 pitch_lag_frac + (pitch_lag_frac > 0 ? 0 : 4),

346 LP_ORDER, AMRWB_SFR_SIZE + 1);

347

348 /* Check which pitch signal path should be used

349 * 6k60 and 8k85 modes have the ltp flag set to 0 */

350 if (amr_subframe->ltp) {

351 memcpy(ctx->pitch_vector, exc, AMRWB_SFR_SIZE * sizeof(float));

352 } else {

353 for (i = 0; i < AMRWB_SFR_SIZE; i++)

354 ctx->pitch_vector[i] = 0.18 * exc[i - 1] + 0.64 * exc[i] +

355 0.18 * exc[i + 1];

356 memcpy(exc, ctx->pitch_vector, AMRWB_SFR_SIZE * sizeof(float));

357 }

358 }

359

360 /** Get x bits in the index interval [lsb,lsb+len-1] inclusive */

361 #define BIT_STR(x,lsb,len) (((x) >> (lsb)) & ((1 << (len)) - 1))

362

363 /** Get the bit at specified position */

364 #define BIT_POS(x, p) (((x) >> (p)) & 1)

365

366 /**

367 * The next six functions decode_[i]p_track decode exactly i pulses

368 * positions and amplitudes (-1 or 1) in a subframe track using

369 * an encoded pulse indexing (TS 26.190 section 5.8.2).

370 *

371 * The results are given in out[], in which a negative number means

372 * amplitude -1 and vice versa (i.e., ampl(x) = x / abs(x) ).

373 *

374 * @param[out] out Output buffer (writes i elements)

375 * @param[in] code Pulse index (no. of bits varies, see below)

376 * @param[in] m (log2) Number of potential positions

377 * @param[in] off Offset for decoded positions

378 */

379 static inline void decode_1p_track(int *out, int code, int m, int off)

380 {

381 int pos = BIT_STR(code, 0, m) + off; ///code: m+1 bits

382

383 out[0] = BIT_POS(code, m) ? -pos : pos;

384 }

385

386 static inline void decode_2p_track(int *out, int code, int m, int off) ///code: 2m+1 bits

387 {

388 int pos0 = BIT_STR(code, m, m) + off;

389 int pos1 = BIT_STR(code, 0, m) + off;

390

391 out[0] = BIT_POS(code, 2*m) ? -pos0 : pos0;

392 out[1] = BIT_POS(code, 2*m) ? -pos1 : pos1;

393 out[1] = pos0 > pos1 ? -out[1] : out[1];

394 }

395

396 static void decode_3p_track(int *out, int code, int m, int off) ///code: 3m+1 bits

397 {

398 int half_2p = BIT_POS(code, 2*m - 1) << (m - 1);

399

400 decode_2p_track(out, BIT_STR(code, 0, 2*m - 1),

401 m - 1, off + half_2p);

402 decode_1p_track(out + 2, BIT_STR(code, 2*m, m + 1), m, off);

403 }

404

405 static void decode_4p_track(int *out, int code, int m, int off) ///code: 4m bits

406 {

407 int half_4p, subhalf_2p;

408 int b_offset = 1 << (m - 1);

409

410 switch (BIT_STR(code, 4*m - 2, 2)) { /* case ID (2 bits) */

411 case 0: /* 0 pulses in A, 4 pulses in B or vice versa */

412 half_4p = BIT_POS(code, 4*m - 3) << (m - 1); // which has 4 pulses

413 subhalf_2p = BIT_POS(code, 2*m - 3) << (m - 2);

414

415 decode_2p_track(out, BIT_STR(code, 0, 2*m - 3),

416 m - 2, off + half_4p + subhalf_2p);

417 decode_2p_track(out + 2, BIT_STR(code, 2*m - 2, 2*m - 1),

418 m - 1, off + half_4p);

419 break;

420 case 1: /* 1 pulse in A, 3 pulses in B */

421 decode_1p_track(out, BIT_STR(code, 3*m - 2, m),

422 m - 1, off);

423 decode_3p_track(out + 1, BIT_STR(code, 0, 3*m - 2),

424 m - 1, off + b_offset);

425 break;

426 case 2: /* 2 pulses in each half */

427 decode_2p_track(out, BIT_STR(code, 2*m - 1, 2*m - 1),

428 m - 1, off);

429 decode_2p_track(out + 2, BIT_STR(code, 0, 2*m - 1),

430 m - 1, off + b_offset);

431 break;

432 case 3: /* 3 pulses in A, 1 pulse in B */

433 decode_3p_track(out, BIT_STR(code, m, 3*m - 2),

434 m - 1, off);

435 decode_1p_track(out + 3, BIT_STR(code, 0, m),

436 m - 1, off + b_offset);

437 break;

438 }

439 }

440

441 static void decode_5p_track(int *out, int code, int m, int off) ///code: 5m bits

442 {

443 int half_3p = BIT_POS(code, 5*m - 1) << (m - 1);

444

445 decode_3p_track(out, BIT_STR(code, 2*m + 1, 3*m - 2),

446 m - 1, off + half_3p);

447

448 decode_2p_track(out + 3, BIT_STR(code, 0, 2*m + 1), m, off);

449 }

450

451 static void decode_6p_track(int *out, int code, int m, int off) ///code: 6m-2 bits

452 {

453 int b_offset = 1 << (m - 1);

454 /* which half has more pulses in cases 0 to 2 */

455 int half_more = BIT_POS(code, 6*m - 5) << (m - 1);

456 int half_other = b_offset - half_more;

457

458 switch (BIT_STR(code, 6*m - 4, 2)) { /* case ID (2 bits) */

459 case 0: /* 0 pulses in A, 6 pulses in B or vice versa */

460 decode_1p_track(out, BIT_STR(code, 0, m),

461 m - 1, off + half_more);

462 decode_5p_track(out + 1, BIT_STR(code, m, 5*m - 5),

463 m - 1, off + half_more);

464 break;

465 case 1: /* 1 pulse in A, 5 pulses in B or vice versa */

466 decode_1p_track(out, BIT_STR(code, 0, m),

467 m - 1, off + half_other);

468 decode_5p_track(out + 1, BIT_STR(code, m, 5*m - 5),

469 m - 1, off + half_more);

470 break;

471 case 2: /* 2 pulses in A, 4 pulses in B or vice versa */

472 decode_2p_track(out, BIT_STR(code, 0, 2*m - 1),

473 m - 1, off + half_other);

474 decode_4p_track(out + 2, BIT_STR(code, 2*m - 1, 4*m - 4),

475 m - 1, off + half_more);

476 break;

477 case 3: /* 3 pulses in A, 3 pulses in B */

478 decode_3p_track(out, BIT_STR(code, 3*m - 2, 3*m - 2),

479 m - 1, off);

480 decode_3p_track(out + 3, BIT_STR(code, 0, 3*m - 2),

481 m - 1, off + b_offset);

482 break;

483 }

484 }

485

486 /**

487 * Decode the algebraic codebook index to pulse positions and signs,

488 * then construct the algebraic codebook vector.

489 *

490 * @param[out] fixed_vector Buffer for the fixed codebook excitation

491 * @param[in] pulse_hi MSBs part of the pulse index array (higher modes only)

492 * @param[in] pulse_lo LSBs part of the pulse index array

493 * @param[in] mode Mode of the current frame

494 */

495 static void decode_fixed_vector(float *fixed_vector, const uint16_t *pulse_hi,

496 const uint16_t *pulse_lo, const enum Mode mode)

497 {

498 /* sig_pos stores for each track the decoded pulse position indexes

499 * (1-based) multiplied by its corresponding amplitude (+1 or -1) */

500 int sig_pos[4][6];

501 int spacing = (mode == MODE_6k60) ? 2 : 4;

502 int i, j;

503

504 switch (mode) {

505 case MODE_6k60:

506 for (i = 0; i < 2; i++)

507 decode_1p_track(sig_pos[i], pulse_lo[i], 5, 1);

508 break;

509 case MODE_8k85:

510 for (i = 0; i < 4; i++)

511 decode_1p_track(sig_pos[i], pulse_lo[i], 4, 1);

512 break;

513 case MODE_12k65:

514 for (i = 0; i < 4; i++)

515 decode_2p_track(sig_pos[i], pulse_lo[i], 4, 1);

516 break;

517 case MODE_14k25:

518 for (i = 0; i < 2; i++)

519 decode_3p_track(sig_pos[i], pulse_lo[i], 4, 1);

520 for (i = 2; i < 4; i++)

521 decode_2p_track(sig_pos[i], pulse_lo[i], 4, 1);

522 break;

523 case MODE_15k85:

524 for (i = 0; i < 4; i++)

525 decode_3p_track(sig_pos[i], pulse_lo[i], 4, 1);

526 break;

527 case MODE_18k25:

528 for (i = 0; i < 4; i++)

529 decode_4p_track(sig_pos[i], (int) pulse_lo[i] +

530 ((int) pulse_hi[i] << 14), 4, 1);

531 break;

532 case MODE_19k85:

533 for (i = 0; i < 2; i++)

534 decode_5p_track(sig_pos[i], (int) pulse_lo[i] +

535 ((int) pulse_hi[i] << 10), 4, 1);

536 for (i = 2; i < 4; i++)

537 decode_4p_track(sig_pos[i], (int) pulse_lo[i] +

538 ((int) pulse_hi[i] << 14), 4, 1);

539 break;

540 case MODE_23k05:

541 case MODE_23k85:

542 for (i = 0; i < 4; i++)

543 decode_6p_track(sig_pos[i], (int) pulse_lo[i] +

544 ((int) pulse_hi[i] << 11), 4, 1);

545 break;

546 }

547

548 memset(fixed_vector, 0, sizeof(float) * AMRWB_SFR_SIZE);

549

550 for (i = 0; i < 4; i++)

551 for (j = 0; j < pulses_nb_per_mode_tr[mode][i]; j++) {

552 int pos = (FFABS(sig_pos[i][j]) - 1) * spacing + i;

553

554 fixed_vector[pos] += sig_pos[i][j] < 0 ? -1.0 : 1.0;

555 }

556 }

557

558 /**

559 * Decode pitch gain and fixed gain correction factor.

560 *

561 * @param[in] vq_gain Vector-quantized index for gains

562 * @param[in] mode Mode of the current frame

563 * @param[out] fixed_gain_factor Decoded fixed gain correction factor

564 * @param[out] pitch_gain Decoded pitch gain

565 */

566 static void decode_gains(const uint8_t vq_gain, const enum Mode mode,

567 float *fixed_gain_factor, float *pitch_gain)

568 {

569 const int16_t *gains = (mode <= MODE_8k85 ? qua_gain_6b[vq_gain] :

570 qua_gain_7b[vq_gain]);

571

572 *pitch_gain = gains[0] * (1.0f / (1 << 14));

573 *fixed_gain_factor = gains[1] * (1.0f / (1 << 11));

574 }

575

576 /**

577 * Apply pitch sharpening filters to the fixed codebook vector.

578 *

579 * @param[in] ctx The context

580 * @param[in,out] fixed_vector Fixed codebook excitation

581 */

582 // XXX: Spec states this procedure should be applied when the pitch

583 // lag is less than 64, but this checking seems absent in reference and AMR-NB

584 static void pitch_sharpening(AMRWBContext *ctx, float *fixed_vector)

585 {

586 int i;

587

588 /* Tilt part */

589 for (i = AMRWB_SFR_SIZE - 1; i != 0; i--)

590 fixed_vector[i] -= fixed_vector[i - 1] * ctx->tilt_coef;

591

592 /* Periodicity enhancement part */

593 for (i = ctx->pitch_lag_int; i < AMRWB_SFR_SIZE; i++)

594 fixed_vector[i] += fixed_vector[i - ctx->pitch_lag_int] * 0.85;

595 }

596

597 /**

598 * Calculate the voicing factor (-1.0 = unvoiced to 1.0 = voiced).

599 *

600 * @param[in] p_vector, f_vector Pitch and fixed excitation vectors

601 * @param[in] p_gain, f_gain Pitch and fixed gains

602 * @param[in] ctx The context

603 */

604 // XXX: There is something wrong with the precision here! The magnitudes

605 // of the energies are not correct. Please check the reference code carefully

606 static float voice_factor(float *p_vector, float p_gain,

607 float *f_vector, float f_gain,

608 CELPMContext *ctx)

609 {

610 double p_ener = (double) ctx->dot_productf(p_vector, p_vector,

611 AMRWB_SFR_SIZE) *

612 p_gain * p_gain;

613 double f_ener = (double) ctx->dot_productf(f_vector, f_vector,

614 AMRWB_SFR_SIZE) *

615 f_gain * f_gain;

616

617 return (p_ener - f_ener) / (p_ener + f_ener);

618 }

619

620 /**

621 * Reduce fixed vector sparseness by smoothing with one of three IR filters,

622 * also known as "adaptive phase dispersion".

623 *

624 * @param[in] ctx The context

625 * @param[in,out] fixed_vector Unfiltered fixed vector

626 * @param[out] buf Space for modified vector if necessary

627 *

628 * @return The potentially overwritten filtered fixed vector address

629 */

630 static float *anti_sparseness(AMRWBContext *ctx,

631 float *fixed_vector, float *buf)

632 {

633 int ir_filter_nr;

634

635 if (ctx->fr_cur_mode > MODE_8k85) // no filtering in higher modes

636 return fixed_vector;

637

638 if (ctx->pitch_gain[0] < 0.6) {

639 ir_filter_nr = 0; // strong filtering

640 } else if (ctx->pitch_gain[0] < 0.9) {

641 ir_filter_nr = 1; // medium filtering

642 } else

643 ir_filter_nr = 2; // no filtering

644

645 /* detect 'onset' */

646 if (ctx->fixed_gain[0] > 3.0 * ctx->fixed_gain[1]) {

647 if (ir_filter_nr < 2)

648 ir_filter_nr++;

649 } else {

650 int i, count = 0;

651

652 for (i = 0; i < 6; i++)

653 if (ctx->pitch_gain[i] < 0.6)

654 count++;

655

656 if (count > 2)

657 ir_filter_nr = 0;

658

659 if (ir_filter_nr > ctx->prev_ir_filter_nr + 1)

660 ir_filter_nr--;

661 }

662

663 /* update ir filter strength history */

664 ctx->prev_ir_filter_nr = ir_filter_nr;

665

666 ir_filter_nr += (ctx->fr_cur_mode == MODE_8k85);

667

668 if (ir_filter_nr < 2) {

669 int i;

670 const float *coef = ir_filters_lookup[ir_filter_nr];

671

672 /* Circular convolution code in the reference

673 * decoder was modified to avoid using one

674 * extra array. The filtered vector is given by:

675 *

676 * c2(n) = sum(i,0,len-1){ c(i) * coef( (n - i + len) % len ) }

677 */

678

679 memset(buf, 0, sizeof(float) * AMRWB_SFR_SIZE);

680 for (i = 0; i < AMRWB_SFR_SIZE; i++)

681 if (fixed_vector[i])

682 ff_celp_circ_addf(buf, buf, coef, i, fixed_vector[i],

683 AMRWB_SFR_SIZE);

684 fixed_vector = buf;

685 }

686

687 return fixed_vector;

688 }

689

690 /**

691 * Calculate a stability factor {teta} based on distance between

692 * current and past isf. A value of 1 shows maximum signal stability.

693 */

694 static float stability_factor(const float *isf, const float *isf_past)

695 {

696 int i;

697 float acc = 0.0;

698

699 for (i = 0; i < LP_ORDER - 1; i++)

700 acc += (isf[i] - isf_past[i]) * (isf[i] - isf_past[i]);

701

702 // XXX: This part is not so clear from the reference code

703 // the result is more accurate changing the "/ 256" to "* 512"

704 return FFMAX(0.0, 1.25 - acc * 0.8 * 512);

705 }

706

707 /**

708 * Apply a non-linear fixed gain smoothing in order to reduce

709 * fluctuation in the energy of excitation.

710 *

711 * @param[in] fixed_gain Unsmoothed fixed gain

712 * @param[in,out] prev_tr_gain Previous threshold gain (updated)

713 * @param[in] voice_fac Frame voicing factor

714 * @param[in] stab_fac Frame stability factor

715 *

716 * @return The smoothed gain

717 */

718 static float noise_enhancer(float fixed_gain, float *prev_tr_gain,

719 float voice_fac, float stab_fac)

720 {

721 float sm_fac = 0.5 * (1 - voice_fac) * stab_fac;

722 float g0;

723

724 // XXX: the following fixed-point constants used to in(de)crement

725 // gain by 1.5dB were taken from the reference code, maybe it could

726 // be simpler

727 if (fixed_gain < *prev_tr_gain) {

728 g0 = FFMIN(*prev_tr_gain, fixed_gain + fixed_gain *

729 (6226 * (1.0f / (1 << 15)))); // +1.5 dB

730 } else

731 g0 = FFMAX(*prev_tr_gain, fixed_gain *

732 (27536 * (1.0f / (1 << 15)))); // -1.5 dB

733

734 *prev_tr_gain = g0; // update next frame threshold

735

736 return sm_fac * g0 + (1 - sm_fac) * fixed_gain;

737 }

738

739 /**

740 * Filter the fixed_vector to emphasize the higher frequencies.

741 *

742 * @param[in,out] fixed_vector Fixed codebook vector

743 * @param[in] voice_fac Frame voicing factor

744 */

745 static void pitch_enhancer(float *fixed_vector, float voice_fac)

746 {

747 int i;

748 float cpe = 0.125 * (1 + voice_fac);

749 float last = fixed_vector[0]; // holds c(i - 1)

750

751 fixed_vector[0] -= cpe * fixed_vector[1];

752

753 for (i = 1; i < AMRWB_SFR_SIZE - 1; i++) {

754 float cur = fixed_vector[i];

755

756 fixed_vector[i] -= cpe * (last + fixed_vector[i + 1]);

757 last = cur;

758 }

759

760 fixed_vector[AMRWB_SFR_SIZE - 1] -= cpe * last;

761 }

762

763 /**

764 * Conduct 16th order linear predictive coding synthesis from excitation.

765 *

766 * @param[in] ctx Pointer to the AMRWBContext

767 * @param[in] lpc Pointer to the LPC coefficients

768 * @param[out] excitation Buffer for synthesis final excitation

769 * @param[in] fixed_gain Fixed codebook gain for synthesis

770 * @param[in] fixed_vector Algebraic codebook vector

771 * @param[in,out] samples Pointer to the output samples and memory

772 */

773 static void synthesis(AMRWBContext *ctx, float *lpc, float *excitation,

774 float fixed_gain, const float *fixed_vector,

775 float *samples)

776 {

777 ctx->acelpv_ctx.weighted_vector_sumf(excitation, ctx->pitch_vector, fixed_vector,

778 ctx->pitch_gain[0], fixed_gain, AMRWB_SFR_SIZE);

779

780 /* emphasize pitch vector contribution in low bitrate modes */

781 if (ctx->pitch_gain[0] > 0.5 && ctx->fr_cur_mode <= MODE_8k85) {

782 int i;

783 float energy = ctx->celpm_ctx.dot_productf(excitation, excitation,

784 AMRWB_SFR_SIZE);

785

786 // XXX: Weird part in both ref code and spec. A unknown parameter

787 // {beta} seems to be identical to the current pitch gain

788 float pitch_factor = 0.25 * ctx->pitch_gain[0] * ctx->pitch_gain[0];

789

790 for (i = 0; i < AMRWB_SFR_SIZE; i++)

791 excitation[i] += pitch_factor * ctx->pitch_vector[i];

792

793 ff_scale_vector_to_given_sum_of_squares(excitation, excitation,

794 energy, AMRWB_SFR_SIZE);

795 }

796

797 ctx->celpf_ctx.celp_lp_synthesis_filterf(samples, lpc, excitation,

798 AMRWB_SFR_SIZE, LP_ORDER);

799 }

800

801 /**

802 * Apply to synthesis a de-emphasis filter of the form:

803 * H(z) = 1 / (1 - m * z^-1)

804 *

805 * @param[out] out Output buffer

806 * @param[in] in Input samples array with in[-1]

807 * @param[in] m Filter coefficient

808 * @param[in,out] mem State from last filtering

809 */

810 static void de_emphasis(float *out, float *in, float m, float mem[1])

811 {

812 int i;

813

814 out[0] = in[0] + m * mem[0];

815

816 for (i = 1; i < AMRWB_SFR_SIZE; i++)

817 out[i] = in[i] + out[i - 1] * m;

818

819 mem[0] = out[AMRWB_SFR_SIZE - 1];

820 }

821

822 /**

823 * Upsample a signal by 5/4 ratio (from 12.8kHz to 16kHz) using

824 * a FIR interpolation filter. Uses past data from before *in address.

825 *

826 * @param[out] out Buffer for interpolated signal

827 * @param[in] in Current signal data (length 0.8*o_size)

828 * @param[in] o_size Output signal length

829 * @param[in] ctx The context

830 */

831 static void upsample_5_4(float *out, const float *in, int o_size, CELPMContext *ctx)

832 {

833 const float *in0 = in - UPS_FIR_SIZE + 1;

834 int i, j, k;

835 int int_part = 0, frac_part;

836

837 i = 0;

838 for (j = 0; j < o_size / 5; j++) {

839 out[i] = in[int_part];

840 frac_part = 4;

841 i++;

842

843 for (k = 1; k < 5; k++) {

844 out[i] = ctx->dot_productf(in0 + int_part,

845 upsample_fir[4 - frac_part],

846 UPS_MEM_SIZE);

847 int_part++;

848 frac_part--;

849 i++;

850 }

851 }

852 }

853

854 /**

855 * Calculate the high-band gain based on encoded index (23k85 mode) or

856 * on the low-band speech signal and the Voice Activity Detection flag.

857 *

858 * @param[in] ctx The context

859 * @param[in] synth LB speech synthesis at 12.8k

860 * @param[in] hb_idx Gain index for mode 23k85 only

861 * @param[in] vad VAD flag for the frame

862 */

863 static float find_hb_gain(AMRWBContext *ctx, const float *synth,

864 uint16_t hb_idx, uint8_t vad)

865 {

866 int wsp = (vad > 0);

867 float tilt;

868

869 if (ctx->fr_cur_mode == MODE_23k85)

870 return qua_hb_gain[hb_idx] * (1.0f / (1 << 14));

871

872 tilt = ctx->celpm_ctx.dot_productf(synth, synth + 1, AMRWB_SFR_SIZE - 1) /

873 ctx->celpm_ctx.dot_productf(synth, synth, AMRWB_SFR_SIZE);

874

875 /* return gain bounded by [0.1, 1.0] */

876 return av_clipf((1.0 - FFMAX(0.0, tilt)) * (1.25 - 0.25 * wsp), 0.1, 1.0);

877 }

878

879 /**

880 * Generate the high-band excitation with the same energy from the lower

881 * one and scaled by the given gain.

882 *

883 * @param[in] ctx The context

884 * @param[out] hb_exc Buffer for the excitation

885 * @param[in] synth_exc Low-band excitation used for synthesis

886 * @param[in] hb_gain Wanted excitation gain

887 */

888 static void scaled_hb_excitation(AMRWBContext *ctx, float *hb_exc,

889 const float *synth_exc, float hb_gain)

890 {

891 int i;

892 float energy = ctx->celpm_ctx.dot_productf(synth_exc, synth_exc,

893 AMRWB_SFR_SIZE);

894

895 /* Generate a white-noise excitation */

896 for (i = 0; i < AMRWB_SFR_SIZE_16k; i++)

897 hb_exc[i] = 32768.0 - (uint16_t) av_lfg_get(&ctx->prng);

898

899 ff_scale_vector_to_given_sum_of_squares(hb_exc, hb_exc,

900 energy * hb_gain * hb_gain,

901 AMRWB_SFR_SIZE_16k);

902 }

903

904 /**

905 * Calculate the auto-correlation for the ISF difference vector.

906 */

907 static float auto_correlation(float *diff_isf, float mean, int lag)

908 {

909 int i;

910 float sum = 0.0;

911

912 for (i = 7; i < LP_ORDER - 2; i++) {

913 float prod = (diff_isf[i] - mean) * (diff_isf[i - lag] - mean);

914 sum += prod * prod;

915 }

916 return sum;

917 }

918

919 /**

920 * Extrapolate a ISF vector to the 16kHz range (20th order LP)

921 * used at mode 6k60 LP filter for the high frequency band.

922 *

923 * @param[out] isf Buffer for extrapolated isf; contains LP_ORDER

924 * values on input

925 */

926 static void extrapolate_isf(float isf[LP_ORDER_16k])

927 {

928 float diff_isf[LP_ORDER - 2], diff_mean;

929 float corr_lag[3];

930 float est, scale;

931 int i, j, i_max_corr;

932

933 isf[LP_ORDER_16k - 1] = isf[LP_ORDER - 1];

934

935 /* Calculate the difference vector */

936 for (i = 0; i < LP_ORDER - 2; i++)

937 diff_isf[i] = isf[i + 1] - isf[i];

938

939 diff_mean = 0.0;

940 for (i = 2; i < LP_ORDER - 2; i++)

941 diff_mean += diff_isf[i] * (1.0f / (LP_ORDER - 4));

942

943 /* Find which is the maximum autocorrelation */

944 i_max_corr = 0;

945 for (i = 0; i < 3; i++) {

946 corr_lag[i] = auto_correlation(diff_isf, diff_mean, i + 2);

947

948 if (corr_lag[i] > corr_lag[i_max_corr])

949 i_max_corr = i;

950 }

951 i_max_corr++;

952

953 for (i = LP_ORDER - 1; i < LP_ORDER_16k - 1; i++)

954 isf[i] = isf[i - 1] + isf[i - 1 - i_max_corr]

955 - isf[i - 2 - i_max_corr];

956

957 /* Calculate an estimate for ISF(18) and scale ISF based on the error */

958 est = 7965 + (isf[2] - isf[3] - isf[4]) / 6.0;

959 scale = 0.5 * (FFMIN(est, 7600) - isf[LP_ORDER - 2]) /

960 (isf[LP_ORDER_16k - 2] - isf[LP_ORDER - 2]);

961

962 for (i = LP_ORDER - 1, j = 0; i < LP_ORDER_16k - 1; i++, j++)

963 diff_isf[j] = scale * (isf[i] - isf[i - 1]);

964

965 /* Stability insurance */

966 for (i = 1; i < LP_ORDER_16k - LP_ORDER; i++)

967 if (diff_isf[i] + diff_isf[i - 1] < 5.0) {

968 if (diff_isf[i] > diff_isf[i - 1]) {

969 diff_isf[i - 1] = 5.0 - diff_isf[i];

970 } else

971 diff_isf[i] = 5.0 - diff_isf[i - 1];

972 }

973

974 for (i = LP_ORDER - 1, j = 0; i < LP_ORDER_16k - 1; i++, j++)

975 isf[i] = isf[i - 1] + diff_isf[j] * (1.0f / (1 << 15));

976

977 /* Scale the ISF vector for 16000 Hz */

978 for (i = 0; i < LP_ORDER_16k - 1; i++)

979 isf[i] *= 0.8;

980 }

981

982 /**

983 * Spectral expand the LP coefficients using the equation:

984 * y[i] = x[i] * (gamma ** i)

985 *

986 * @param[out] out Output buffer (may use input array)

987 * @param[in] lpc LP coefficients array

988 * @param[in] gamma Weighting factor

989 * @param[in] size LP array size

990 */

991 static void lpc_weighting(float *out, const float *lpc, float gamma, int size)

992 {

993 int i;

994 float fac = gamma;

995

996 for (i = 0; i < size; i++) {

997 out[i] = lpc[i] * fac;

998 fac *= gamma;

999 }

1000 }

1001

1002 /**

1003 * Conduct 20th order linear predictive coding synthesis for the high

1004 * frequency band excitation at 16kHz.

1005 *

1006 * @param[in] ctx The context

1007 * @param[in] subframe Current subframe index (0 to 3)

1008 * @param[in,out] samples Pointer to the output speech samples

1009 * @param[in] exc Generated white-noise scaled excitation

1010 * @param[in] isf Current frame isf vector

1011 * @param[in] isf_past Past frame final isf vector

1012 */

1013 static void hb_synthesis(AMRWBContext *ctx, int subframe, float *samples,

1014 const float *exc, const float *isf, const float *isf_past)

1015 {

1016 float hb_lpc[LP_ORDER_16k];

1017 enum Mode mode = ctx->fr_cur_mode;

1018

1019 if (mode == MODE_6k60) {

1020 float e_isf[LP_ORDER_16k]; // ISF vector for extrapolation

1021 double e_isp[LP_ORDER_16k];

1022

1023 ctx->acelpv_ctx.weighted_vector_sumf(e_isf, isf_past, isf, isfp_inter[subframe],

1024 1.0 - isfp_inter[subframe], LP_ORDER);

1025

1026 extrapolate_isf(e_isf);

1027

1028 e_isf[LP_ORDER_16k - 1] *= 2.0;

1029 ff_acelp_lsf2lspd(e_isp, e_isf, LP_ORDER_16k);

1030 ff_amrwb_lsp2lpc(e_isp, hb_lpc, LP_ORDER_16k);

1031

1032 lpc_weighting(hb_lpc, hb_lpc, 0.9, LP_ORDER_16k);

1033 } else {

1034 lpc_weighting(hb_lpc, ctx->lp_coef[subframe], 0.6, LP_ORDER);

1035 }

1036

1037 ctx->celpf_ctx.celp_lp_synthesis_filterf(samples, hb_lpc, exc, AMRWB_SFR_SIZE_16k,

1038 (mode == MODE_6k60) ? LP_ORDER_16k : LP_ORDER);

1039 }

1040

1041 /**

1042 * Apply a 15th order filter to high-band samples.

1043 * The filter characteristic depends on the given coefficients.

1044 *

1045 * @param[out] out Buffer for filtered output

1046 * @param[in] fir_coef Filter coefficients

1047 * @param[in,out] mem State from last filtering (updated)

1048 * @param[in] in Input speech data (high-band)

1049 *

1050 * @remark It is safe to pass the same array in in and out parameters

1051 */

1052

1053 #ifndef hb_fir_filter

1054 static void hb_fir_filter(float *out, const float fir_coef[HB_FIR_SIZE + 1],

1055 float mem[HB_FIR_SIZE], const float *in)

1056 {

1057 int i, j;

1058 float data[AMRWB_SFR_SIZE_16k + HB_FIR_SIZE]; // past and current samples

1059

1060 memcpy(data, mem, HB_FIR_SIZE * sizeof(float));

1061 memcpy(data + HB_FIR_SIZE, in, AMRWB_SFR_SIZE_16k * sizeof(float));

1062

1063 for (i = 0; i < AMRWB_SFR_SIZE_16k; i++) {

1064 out[i] = 0.0;

1065 for (j = 0; j <= HB_FIR_SIZE; j++)

1066 out[i] += data[i + j] * fir_coef[j];

1067 }

1068

1069 memcpy(mem, data + AMRWB_SFR_SIZE_16k, HB_FIR_SIZE * sizeof(float));

1070 }

1071 #endif /* hb_fir_filter */

1072

1073 /**

1074 * Update context state before the next subframe.

1075 */

1076 static void update_sub_state(AMRWBContext *ctx)

1077 {

1078 memmove(&ctx->excitation_buf[0], &ctx->excitation_buf[AMRWB_SFR_SIZE],

1079 (AMRWB_P_DELAY_MAX + LP_ORDER + 1) * sizeof(float));

1080

1081 memmove(&ctx->pitch_gain[1], &ctx->pitch_gain[0], 5 * sizeof(float));

1082 memmove(&ctx->fixed_gain[1], &ctx->fixed_gain[0], 1 * sizeof(float));

1083

1084 memmove(&ctx->samples_az[0], &ctx->samples_az[AMRWB_SFR_SIZE],

1085 LP_ORDER * sizeof(float));

1086 memmove(&ctx->samples_up[0], &ctx->samples_up[AMRWB_SFR_SIZE],

1087 UPS_MEM_SIZE * sizeof(float));

1088 memmove(&ctx->samples_hb[0], &ctx->samples_hb[AMRWB_SFR_SIZE_16k],

1089 LP_ORDER_16k * sizeof(float));

1090 }

1091

1092 static int amrwb_decode_frame(AVCodecContext *avctx, void *data,

1093 int *got_frame_ptr, AVPacket *avpkt)

1094 {

1095 AMRWBContext *ctx = avctx->priv_data;

1096 AVFrame *frame = data;

1097 AMRWBFrame *cf = &ctx->frame;

1098 const uint8_t *buf = avpkt->data;

1099 int buf_size = avpkt->size;

1100 int expected_fr_size, header_size;

1101 float *buf_out;

1102 float spare_vector[AMRWB_SFR_SIZE]; // extra stack space to hold result from anti-sparseness processing

1103 float fixed_gain_factor; // fixed gain correction factor (gamma)

1104 float *synth_fixed_vector; // pointer to the fixed vector that synthesis should use

1105 float synth_fixed_gain; // the fixed gain that synthesis should use

1106 float voice_fac, stab_fac; // parameters used for gain smoothing

1107 float synth_exc[AMRWB_SFR_SIZE]; // post-processed excitation for synthesis

1108 float hb_exc[AMRWB_SFR_SIZE_16k]; // excitation for the high frequency band

1109 float hb_samples[AMRWB_SFR_SIZE_16k]; // filtered high-band samples from synthesis

1110 float hb_gain;

1111 int sub, i, ret;

1112

1113 /* get output buffer */

1114 frame->nb_samples = 4 * AMRWB_SFR_SIZE_16k;

1115 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)

1116 return ret;

1117 buf_out = (float *)frame->data[0];

1118

1119 header_size = decode_mime_header(ctx, buf);

1120 if (ctx->fr_cur_mode > MODE_SID) {

1121 av_log(avctx, AV_LOG_ERROR,

1122 "Invalid mode %d\n", ctx->fr_cur_mode);

1123 return AVERROR_INVALIDDATA;

1124 }

1125 expected_fr_size = ((cf_sizes_wb[ctx->fr_cur_mode] + 7) >> 3) + 1;

1126

1127 if (buf_size < expected_fr_size) {

1128 av_log(avctx, AV_LOG_ERROR,

1129 "Frame too small (%d bytes). Truncated file?\n", buf_size);

1130 *got_frame_ptr = 0;

1131 return AVERROR_INVALIDDATA;

1132 }

1133

1134 if (!ctx->fr_quality || ctx->fr_cur_mode > MODE_SID)

1135 av_log(avctx, AV_LOG_ERROR, "Encountered a bad or corrupted frame\n");

1136

1137 if (ctx->fr_cur_mode == MODE_SID) { /* Comfort noise frame */

1138 avpriv_request_sample(avctx, "SID mode");

1139 return AVERROR_PATCHWELCOME;

1140 }

1141

1142 ff_amr_bit_reorder((uint16_t *) &ctx->frame, sizeof(AMRWBFrame),

1143 buf + header_size, amr_bit_orderings_by_mode[ctx->fr_cur_mode]);

1144

1145 /* Decode the quantized ISF vector */

1146 if (ctx->fr_cur_mode == MODE_6k60) {

1147 decode_isf_indices_36b(cf->isp_id, ctx->isf_cur);

1148 } else {

1149 decode_isf_indices_46b(cf->isp_id, ctx->isf_cur);

1150 }

1151

1152 isf_add_mean_and_past(ctx->isf_cur, ctx->isf_q_past);

1153 ff_set_min_dist_lsf(ctx->isf_cur, MIN_ISF_SPACING, LP_ORDER - 1);

1154

1155 stab_fac = stability_factor(ctx->isf_cur, ctx->isf_past_final);

1156

1157 ctx->isf_cur[LP_ORDER - 1] *= 2.0;

1158 ff_acelp_lsf2lspd(ctx->isp[3], ctx->isf_cur, LP_ORDER);

1159

1160 /* Generate a ISP vector for each subframe */

1161 if (ctx->first_frame) {

1162 ctx->first_frame = 0;

1163 memcpy(ctx->isp_sub4_past, ctx->isp[3], LP_ORDER * sizeof(double));

1164 }

1165 interpolate_isp(ctx->isp, ctx->isp_sub4_past);

1166

1167 for (sub = 0; sub < 4; sub++)

1168 ff_amrwb_lsp2lpc(ctx->isp[sub], ctx->lp_coef[sub], LP_ORDER);

1169

1170 for (sub = 0; sub < 4; sub++) {

1171 const AMRWBSubFrame *cur_subframe = &cf->subframe[sub];

1172 float *sub_buf = buf_out + sub * AMRWB_SFR_SIZE_16k;

1173

1174 /* Decode adaptive codebook (pitch vector) */

1175 decode_pitch_vector(ctx, cur_subframe, sub);

1176 /* Decode innovative codebook (fixed vector) */

1177 decode_fixed_vector(ctx->fixed_vector, cur_subframe->pul_ih,

1178 cur_subframe->pul_il, ctx->fr_cur_mode);

1179

1180 pitch_sharpening(ctx, ctx->fixed_vector);

1181

1182 decode_gains(cur_subframe->vq_gain, ctx->fr_cur_mode,

1183 &fixed_gain_factor, &ctx->pitch_gain[0]);

1184

1185 ctx->fixed_gain[0] =

1186 ff_amr_set_fixed_gain(fixed_gain_factor,

1187 ctx->celpm_ctx.dot_productf(ctx->fixed_vector,

1188 ctx->fixed_vector,

1189 AMRWB_SFR_SIZE) /

1190 AMRWB_SFR_SIZE,

1191 ctx->prediction_error,

1192 ENERGY_MEAN, energy_pred_fac);

1193

1194 /* Calculate voice factor and store tilt for next subframe */

1195 voice_fac = voice_factor(ctx->pitch_vector, ctx->pitch_gain[0],

1196 ctx->fixed_vector, ctx->fixed_gain[0],

1197 &ctx->celpm_ctx);

1198 ctx->tilt_coef = voice_fac * 0.25 + 0.25;

1199

1200 /* Construct current excitation */

1201 for (i = 0; i < AMRWB_SFR_SIZE; i++) {

1202 ctx->excitation[i] *= ctx->pitch_gain[0];

1203 ctx->excitation[i] += ctx->fixed_gain[0] * ctx->fixed_vector[i];

1204 ctx->excitation[i] = truncf(ctx->excitation[i]);

1205 }

1206

1207 /* Post-processing of excitation elements */

1208 synth_fixed_gain = noise_enhancer(ctx->fixed_gain[0], &ctx->prev_tr_gain,

1209 voice_fac, stab_fac);

1210

1211 synth_fixed_vector = anti_sparseness(ctx, ctx->fixed_vector,

1212 spare_vector);

1213

1214 pitch_enhancer(synth_fixed_vector, voice_fac);

1215

1216 synthesis(ctx, ctx->lp_coef[sub], synth_exc, synth_fixed_gain,

1217 synth_fixed_vector, &ctx->samples_az[LP_ORDER]);

1218

1219 /* Synthesis speech post-processing */

1220 de_emphasis(&ctx->samples_up[UPS_MEM_SIZE],

1221 &ctx->samples_az[LP_ORDER], PREEMPH_FAC, ctx->demph_mem);

1222

1223 ctx->acelpf_ctx.acelp_apply_order_2_transfer_function(&ctx->samples_up[UPS_MEM_SIZE],

1224 &ctx->samples_up[UPS_MEM_SIZE], hpf_zeros, hpf_31_poles,

1225 hpf_31_gain, ctx->hpf_31_mem, AMRWB_SFR_SIZE);

1226

1227 upsample_5_4(sub_buf, &ctx->samples_up[UPS_FIR_SIZE],

1228 AMRWB_SFR_SIZE_16k, &ctx->celpm_ctx);

1229

1230 /* High frequency band (6.4 - 7.0 kHz) generation part */

1231 ctx->acelpf_ctx.acelp_apply_order_2_transfer_function(hb_samples,

1232 &ctx->samples_up[UPS_MEM_SIZE], hpf_zeros, hpf_400_poles,

1233 hpf_400_gain, ctx->hpf_400_mem, AMRWB_SFR_SIZE);

1234

1235 hb_gain = find_hb_gain(ctx, hb_samples,

1236 cur_subframe->hb_gain, cf->vad);

1237

1238 scaled_hb_excitation(ctx, hb_exc, synth_exc, hb_gain);

1239

1240 hb_synthesis(ctx, sub, &ctx->samples_hb[LP_ORDER_16k],

1241 hb_exc, ctx->isf_cur, ctx->isf_past_final);

1242

1243 /* High-band post-processing filters */

1244 hb_fir_filter(hb_samples, bpf_6_7_coef, ctx->bpf_6_7_mem,

1245 &ctx->samples_hb[LP_ORDER_16k]);

1246

1247 if (ctx->fr_cur_mode == MODE_23k85)

1248 hb_fir_filter(hb_samples, lpf_7_coef, ctx->lpf_7_mem,

1249 hb_samples);

1250

1251 /* Add the low and high frequency bands */

1252 for (i = 0; i < AMRWB_SFR_SIZE_16k; i++)

1253 sub_buf[i] = (sub_buf[i] + hb_samples[i]) * (1.0f / (1 << 15));

1254

1255 /* Update buffers and history */

1256 update_sub_state(ctx);

1257 }

1258

1259 /* update state for next frame */

1260 memcpy(ctx->isp_sub4_past, ctx->isp[3], LP_ORDER * sizeof(ctx->isp[3][0]));

1261 memcpy(ctx->isf_past_final, ctx->isf_cur, LP_ORDER * sizeof(float));

1262

1263 *got_frame_ptr = 1;

1264

1265 return expected_fr_size;

1266 }

1267

1268 AVCodec ff_amrwb_decoder = {

1269 .name = "amrwb",

1270 .long_name = NULL_IF_CONFIG_SMALL("AMR-WB (Adaptive Multi-Rate WideBand)"),

1271 .type = AVMEDIA_TYPE_AUDIO,

1272 .id = AV_CODEC_ID_AMR_WB,

1273 .priv_data_size = sizeof(AMRWBContext),

1274 .init = amrwb_decode_init,

1275 .decode = amrwb_decode_frame,

1276 .capabilities = CODEC_CAP_DR1,

1277 .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT,

1278 AV_SAMPLE_FMT_NONE },

1279 };

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