FFmpeg: libavcodec/amrnbdec.c Source File

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amrnbdec.c

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1 /*

2 * AMR narrowband decoder

5 *

6 * This file is part of FFmpeg.

7 *

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

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

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

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

12 *

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

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

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

16 * Lesser General Public License for more details.

17 *

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

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

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

21 */

24 /**

25 * @file

26 * AMR narrowband decoder

27 *

28 * This decoder uses floats for simplicity and so is not bit-exact. One

29 * difference is that differences in phase can accumulate. The test sequences

30 * in 3GPP TS 26.074 can still be useful.

31 *

32 * - Comparing this file's output to the output of the ref decoder gives a

33 * PSNR of 30 to 80. Plotting the output samples shows a difference in

34 * phase in some areas.

35 *

36 * - Comparing both decoders against their input, this decoder gives a similar

37 * PSNR. If the test sequence homing frames are removed (this decoder does

38 * not detect them), the PSNR is at least as good as the reference on 140

39 * out of 169 tests.

40 */

43 #include <string.h>

44 #include <math.h>

46 #include "libavutil/channel_layout.h"

47 #include "libavutil/float_dsp.h"

48 #include "avcodec.h"

49 #include "libavutil/common.h"

50 #include "libavutil/avassert.h"

51 #include "celp_math.h"

52 #include "celp_filters.h"

53 #include "acelp_filters.h"

54 #include "acelp_vectors.h"

55 #include "acelp_pitch_delay.h"

56 #include "lsp.h"

57 #include "amr.h"

58 #include "internal.h"

60 #include "amrnbdata.h"

62 #define AMR_BLOCK_SIZE 160 ///< samples per frame

63 #define AMR_SAMPLE_BOUND 32768.0 ///< threshold for synthesis overflow

65 /**

66 * Scale from constructed speech to [-1,1]

67 *

68 * AMR is designed to produce 16-bit PCM samples (3GPP TS 26.090 4.2) but

69 * upscales by two (section 6.2.2).

70 *

71 * Fundamentally, this scale is determined by energy_mean through

72 * the fixed vector contribution to the excitation vector.

73 */

74 #define AMR_SAMPLE_SCALE (2.0 / 32768.0)

76 /** Prediction factor for 12.2kbit/s mode */

77 #define PRED_FAC_MODE_12k2 0.65

79 #define LSF_R_FAC (8000.0 / 32768.0) ///< LSF residual tables to Hertz

80 #define MIN_LSF_SPACING (50.0488 / 8000.0) ///< Ensures stability of LPC filter

81 #define PITCH_LAG_MIN_MODE_12k2 18 ///< Lower bound on decoded lag search in 12.2kbit/s mode

83 /** Initial energy in dB. Also used for bad frames (unimplemented). */

84 #define MIN_ENERGY -14.0

86 /** Maximum sharpening factor

87 *

88 * The specification says 0.8, which should be 13107, but the reference C code

89 * uses 13017 instead. (Amusingly the same applies to SHARP_MAX in g729dec.c.)

90 */

91 #define SHARP_MAX 0.79449462890625

93 /** Number of impulse response coefficients used for tilt factor */

94 #define AMR_TILT_RESPONSE 22

95 /** Tilt factor = 1st reflection coefficient * gamma_t */

96 #define AMR_TILT_GAMMA_T 0.8

97 /** Adaptive gain control factor used in post-filter */

98 #define AMR_AGC_ALPHA 0.9

100 typedef struct AMRContext {

101 AMRNBFrame frame; ///< decoded AMR parameters (lsf coefficients, codebook indexes, etc)

102 uint8_t bad_frame_indicator; ///< bad frame ? 1 : 0

103 enum Mode cur_frame_mode;

104

105 int16_t prev_lsf_r[LP_FILTER_ORDER]; ///< residual LSF vector from previous subframe

106 double lsp[4][LP_FILTER_ORDER]; ///< lsp vectors from current frame

107 double prev_lsp_sub4[LP_FILTER_ORDER]; ///< lsp vector for the 4th subframe of the previous frame

108

109 float lsf_q[4][LP_FILTER_ORDER]; ///< Interpolated LSF vector for fixed gain smoothing

110 float lsf_avg[LP_FILTER_ORDER]; ///< vector of averaged lsf vector

111

112 float lpc[4][LP_FILTER_ORDER]; ///< lpc coefficient vectors for 4 subframes

113

114 uint8_t pitch_lag_int; ///< integer part of pitch lag from current subframe

115

116 float excitation_buf[PITCH_DELAY_MAX + LP_FILTER_ORDER + 1 + AMR_SUBFRAME_SIZE]; ///< current excitation and all necessary excitation history

117 float *excitation; ///< pointer to the current excitation vector in excitation_buf

118

119 float pitch_vector[AMR_SUBFRAME_SIZE]; ///< adaptive code book (pitch) vector

120 float fixed_vector[AMR_SUBFRAME_SIZE]; ///< algebraic codebook (fixed) vector (must be kept zero between frames)

121

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

123 float pitch_gain[5]; ///< quantified pitch gains for the current and previous four subframes

124 float fixed_gain[5]; ///< quantified fixed gains for the current and previous four subframes

125

126 float beta; ///< previous pitch_gain, bounded by [0.0,SHARP_MAX]

127 uint8_t diff_count; ///< the number of subframes for which diff has been above 0.65

128 uint8_t hang_count; ///< the number of subframes since a hangover period started

129

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

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

132 uint8_t ir_filter_onset; ///< flag for impulse response filter strength

133

134 float postfilter_mem[10]; ///< previous intermediate values in the formant filter

135 float tilt_mem; ///< previous input to tilt compensation filter

136 float postfilter_agc; ///< previous factor used for adaptive gain control

137 float high_pass_mem[2]; ///< previous intermediate values in the high-pass filter

138

139 float samples_in[LP_FILTER_ORDER + AMR_SUBFRAME_SIZE]; ///< floating point samples

140

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

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

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

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

145

146 } AMRContext;

147

148 /** Double version of ff_weighted_vector_sumf() */

149 static void weighted_vector_sumd(double *out, const double *in_a,

150 const double *in_b, double weight_coeff_a,

151 double weight_coeff_b, int length)

152 {

153 int i;

154

155 for (i = 0; i < length; i++)

156 out[i] = weight_coeff_a * in_a[i]

157 + weight_coeff_b * in_b[i];

158 }

159

160 static av_cold int amrnb_decode_init(AVCodecContext *avctx)

161 {

162 AMRContext *p = avctx->priv_data;

163 int i;

164

165 if (avctx->channels > 1) {

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

167 return AVERROR_PATCHWELCOME;

168 }

169

170 avctx->channels = 1;

171 avctx->channel_layout = AV_CH_LAYOUT_MONO;

172 if (!avctx->sample_rate)

173 avctx->sample_rate = 8000;

174 avctx->sample_fmt = AV_SAMPLE_FMT_FLT;

175

176 // p->excitation always points to the same position in p->excitation_buf

177 p->excitation = &p->excitation_buf[PITCH_DELAY_MAX + LP_FILTER_ORDER + 1];

178

179 for (i = 0; i < LP_FILTER_ORDER; i++) {

180 p->prev_lsp_sub4[i] = lsp_sub4_init[i] * 1000 / (float)(1 << 15);

181 p->lsf_avg[i] = p->lsf_q[3][i] = lsp_avg_init[i] / (float)(1 << 15);

182 }

183

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

185 p->prediction_error[i] = MIN_ENERGY;

186

187 ff_acelp_filter_init(&p->acelpf_ctx);

188 ff_acelp_vectors_init(&p->acelpv_ctx);

189 ff_celp_filter_init(&p->celpf_ctx);

190 ff_celp_math_init(&p->celpm_ctx);

191

192 return 0;

193 }

194

195

196 /**

197 * Unpack an RFC4867 speech frame into the AMR frame mode and parameters.

198 *

199 * The order of speech bits is specified by 3GPP TS 26.101.

200 *

201 * @param p the context

202 * @param buf pointer to the input buffer

203 * @param buf_size size of the input buffer

204 *

205 * @return the frame mode

206 */

207 static enum Mode unpack_bitstream(AMRContext *p, const uint8_t *buf,

208 int buf_size)

209 {

210 enum Mode mode;

211

212 // Decode the first octet.

213 mode = buf[0] >> 3 & 0x0F; // frame type

214 p->bad_frame_indicator = (buf[0] & 0x4) != 0x4; // quality bit

215

216 if (mode >= N_MODES || buf_size < frame_sizes_nb[mode] + 1) {

217 return NO_DATA;

218 }

219

220 if (mode < MODE_DTX)

221 ff_amr_bit_reorder((uint16_t *) &p->frame, sizeof(AMRNBFrame), buf + 1,

222 amr_unpacking_bitmaps_per_mode[mode]);

223

224 return mode;

225 }

226

227

228 /// @name AMR pitch LPC coefficient decoding functions

229 /// @{

230

231 /**

232 * Interpolate the LSF vector (used for fixed gain smoothing).

233 * The interpolation is done over all four subframes even in MODE_12k2.

234 *

235 * @param[in] ctx The Context

236 * @param[in,out] lsf_q LSFs in [0,1] for each subframe

237 * @param[in] lsf_new New LSFs in [0,1] for subframe 4

238 */

239 static void interpolate_lsf(ACELPVContext *ctx, float lsf_q[4][LP_FILTER_ORDER], float *lsf_new)

240 {

241 int i;

242

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

244 ctx->weighted_vector_sumf(lsf_q[i], lsf_q[3], lsf_new,

245 0.25 * (3 - i), 0.25 * (i + 1),

246 LP_FILTER_ORDER);

247 }

248

249 /**

250 * Decode a set of 5 split-matrix quantized lsf indexes into an lsp vector.

251 *

252 * @param p the context

253 * @param lsp output LSP vector

254 * @param lsf_no_r LSF vector without the residual vector added

255 * @param lsf_quantizer pointers to LSF dictionary tables

256 * @param quantizer_offset offset in tables

257 * @param sign for the 3 dictionary table

258 * @param update store data for computing the next frame's LSFs

259 */

260 static void lsf2lsp_for_mode12k2(AMRContext *p, double lsp[LP_FILTER_ORDER],

261 const float lsf_no_r[LP_FILTER_ORDER],

262 const int16_t *lsf_quantizer[5],

263 const int quantizer_offset,

264 const int sign, const int update)

265 {

266 int16_t lsf_r[LP_FILTER_ORDER]; // residual LSF vector

267 float lsf_q[LP_FILTER_ORDER]; // quantified LSF vector

268 int i;

269

270 for (i = 0; i < LP_FILTER_ORDER >> 1; i++)

271 memcpy(&lsf_r[i << 1], &lsf_quantizer[i][quantizer_offset],

272 2 * sizeof(*lsf_r));

273

274 if (sign) {

275 lsf_r[4] *= -1;

276 lsf_r[5] *= -1;

277 }

278

279 if (update)

280 memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(*lsf_r));

281

282 for (i = 0; i < LP_FILTER_ORDER; i++)

283 lsf_q[i] = lsf_r[i] * (LSF_R_FAC / 8000.0) + lsf_no_r[i] * (1.0 / 8000.0);

284

285 ff_set_min_dist_lsf(lsf_q, MIN_LSF_SPACING, LP_FILTER_ORDER);

286

287 if (update)

288 interpolate_lsf(&p->acelpv_ctx, p->lsf_q, lsf_q);

289

290 ff_acelp_lsf2lspd(lsp, lsf_q, LP_FILTER_ORDER);

291 }

292

293 /**

294 * Decode a set of 5 split-matrix quantized lsf indexes into 2 lsp vectors.

295 *

296 * @param p pointer to the AMRContext

297 */

298 static void lsf2lsp_5(AMRContext *p)

299 {

300 const uint16_t *lsf_param = p->frame.lsf;

301 float lsf_no_r[LP_FILTER_ORDER]; // LSFs without the residual vector

302 const int16_t *lsf_quantizer[5];

303 int i;

304

305 lsf_quantizer[0] = lsf_5_1[lsf_param[0]];

306 lsf_quantizer[1] = lsf_5_2[lsf_param[1]];

307 lsf_quantizer[2] = lsf_5_3[lsf_param[2] >> 1];

308 lsf_quantizer[3] = lsf_5_4[lsf_param[3]];

309 lsf_quantizer[4] = lsf_5_5[lsf_param[4]];

310

311 for (i = 0; i < LP_FILTER_ORDER; i++)

312 lsf_no_r[i] = p->prev_lsf_r[i] * LSF_R_FAC * PRED_FAC_MODE_12k2 + lsf_5_mean[i];

313

314 lsf2lsp_for_mode12k2(p, p->lsp[1], lsf_no_r, lsf_quantizer, 0, lsf_param[2] & 1, 0);

315 lsf2lsp_for_mode12k2(p, p->lsp[3], lsf_no_r, lsf_quantizer, 2, lsf_param[2] & 1, 1);

316

317 // interpolate LSP vectors at subframes 1 and 3

318 weighted_vector_sumd(p->lsp[0], p->prev_lsp_sub4, p->lsp[1], 0.5, 0.5, LP_FILTER_ORDER);

319 weighted_vector_sumd(p->lsp[2], p->lsp[1] , p->lsp[3], 0.5, 0.5, LP_FILTER_ORDER);

320 }

321

322 /**

323 * Decode a set of 3 split-matrix quantized lsf indexes into an lsp vector.

324 *

325 * @param p pointer to the AMRContext

326 */

327 static void lsf2lsp_3(AMRContext *p)

328 {

329 const uint16_t *lsf_param = p->frame.lsf;

330 int16_t lsf_r[LP_FILTER_ORDER]; // residual LSF vector

331 float lsf_q[LP_FILTER_ORDER]; // quantified LSF vector

332 const int16_t *lsf_quantizer;

333 int i, j;

334

335 lsf_quantizer = (p->cur_frame_mode == MODE_7k95 ? lsf_3_1_MODE_7k95 : lsf_3_1)[lsf_param[0]];

336 memcpy(lsf_r, lsf_quantizer, 3 * sizeof(*lsf_r));

337

338 lsf_quantizer = lsf_3_2[lsf_param[1] << (p->cur_frame_mode <= MODE_5k15)];

339 memcpy(lsf_r + 3, lsf_quantizer, 3 * sizeof(*lsf_r));

340

341 lsf_quantizer = (p->cur_frame_mode <= MODE_5k15 ? lsf_3_3_MODE_5k15 : lsf_3_3)[lsf_param[2]];

342 memcpy(lsf_r + 6, lsf_quantizer, 4 * sizeof(*lsf_r));

343

344 // calculate mean-removed LSF vector and add mean

345 for (i = 0; i < LP_FILTER_ORDER; i++)

346 lsf_q[i] = (lsf_r[i] + p->prev_lsf_r[i] * pred_fac[i]) * (LSF_R_FAC / 8000.0) + lsf_3_mean[i] * (1.0 / 8000.0);

347

348 ff_set_min_dist_lsf(lsf_q, MIN_LSF_SPACING, LP_FILTER_ORDER);

349

350 // store data for computing the next frame's LSFs

351 interpolate_lsf(&p->acelpv_ctx, p->lsf_q, lsf_q);

352 memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(*lsf_r));

353

354 ff_acelp_lsf2lspd(p->lsp[3], lsf_q, LP_FILTER_ORDER);

355

356 // interpolate LSP vectors at subframes 1, 2 and 3

357 for (i = 1; i <= 3; i++)

358 for(j = 0; j < LP_FILTER_ORDER; j++)

359 p->lsp[i-1][j] = p->prev_lsp_sub4[j] +

360 (p->lsp[3][j] - p->prev_lsp_sub4[j]) * 0.25 * i;

361 }

362

363 /// @}

364

365

366 /// @name AMR pitch vector decoding functions

367 /// @{

368

369 /**

370 * Like ff_decode_pitch_lag(), but with 1/6 resolution

371 */

372 static void decode_pitch_lag_1_6(int *lag_int, int *lag_frac, int pitch_index,

373 const int prev_lag_int, const int subframe)

374 {

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

376 if (pitch_index < 463) {

377 *lag_int = (pitch_index + 107) * 10923 >> 16;

378 *lag_frac = pitch_index - *lag_int * 6 + 105;

379 } else {

380 *lag_int = pitch_index - 368;

381 *lag_frac = 0;

382 }

383 } else {

384 *lag_int = ((pitch_index + 5) * 10923 >> 16) - 1;

385 *lag_frac = pitch_index - *lag_int * 6 - 3;

386 *lag_int += av_clip(prev_lag_int - 5, PITCH_LAG_MIN_MODE_12k2,

387 PITCH_DELAY_MAX - 9);

388 }

389 }

390

391 static void decode_pitch_vector(AMRContext *p,

392 const AMRNBSubframe *amr_subframe,

393 const int subframe)

394 {

395 int pitch_lag_int, pitch_lag_frac;

396 enum Mode mode = p->cur_frame_mode;

397

398 if (p->cur_frame_mode == MODE_12k2) {

399 decode_pitch_lag_1_6(&pitch_lag_int, &pitch_lag_frac,

400 amr_subframe->p_lag, p->pitch_lag_int,

401 subframe);

402 } else

403 ff_decode_pitch_lag(&pitch_lag_int, &pitch_lag_frac,

404 amr_subframe->p_lag,

405 p->pitch_lag_int, subframe,

406 mode != MODE_4k75 && mode != MODE_5k15,

407 mode <= MODE_6k7 ? 4 : (mode == MODE_7k95 ? 5 : 6));

408

409 p->pitch_lag_int = pitch_lag_int; // store previous lag in a uint8_t

410

411 pitch_lag_frac <<= (p->cur_frame_mode != MODE_12k2);

412

413 pitch_lag_int += pitch_lag_frac > 0;

414

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

416 pitch lag using a b60 hamming windowed sinc function. */

417 p->acelpf_ctx.acelp_interpolatef(p->excitation,

418 p->excitation + 1 - pitch_lag_int,

419 ff_b60_sinc, 6,

420 pitch_lag_frac + 6 - 6*(pitch_lag_frac > 0),

421 10, AMR_SUBFRAME_SIZE);

422

423 memcpy(p->pitch_vector, p->excitation, AMR_SUBFRAME_SIZE * sizeof(float));

424 }

425

426 /// @}

427

428

429 /// @name AMR algebraic code book (fixed) vector decoding functions

430 /// @{

431

432 /**

433 * Decode a 10-bit algebraic codebook index from a 10.2 kbit/s frame.

434 */

435 static void decode_10bit_pulse(int code, int pulse_position[8],

436 int i1, int i2, int i3)

437 {

438 // coded using 7+3 bits with the 3 LSBs being, individually, the LSB of 1 of

439 // the 3 pulses and the upper 7 bits being coded in base 5

440 const uint8_t *positions = base_five_table[code >> 3];

441 pulse_position[i1] = (positions[2] << 1) + ( code & 1);

442 pulse_position[i2] = (positions[1] << 1) + ((code >> 1) & 1);

443 pulse_position[i3] = (positions[0] << 1) + ((code >> 2) & 1);

444 }

445

446 /**

447 * Decode the algebraic codebook index to pulse positions and signs and

448 * construct the algebraic codebook vector for MODE_10k2.

449 *

450 * @param fixed_index positions of the eight pulses

451 * @param fixed_sparse pointer to the algebraic codebook vector

452 */

453 static void decode_8_pulses_31bits(const int16_t *fixed_index,

454 AMRFixed *fixed_sparse)

455 {

456 int pulse_position[8];

457 int i, temp;

458

459 decode_10bit_pulse(fixed_index[4], pulse_position, 0, 4, 1);

460 decode_10bit_pulse(fixed_index[5], pulse_position, 2, 6, 5);

461

462 // coded using 5+2 bits with the 2 LSBs being, individually, the LSB of 1 of

463 // the 2 pulses and the upper 5 bits being coded in base 5

464 temp = ((fixed_index[6] >> 2) * 25 + 12) >> 5;

465 pulse_position[3] = temp % 5;

466 pulse_position[7] = temp / 5;

467 if (pulse_position[7] & 1)

468 pulse_position[3] = 4 - pulse_position[3];

469 pulse_position[3] = (pulse_position[3] << 1) + ( fixed_index[6] & 1);

470 pulse_position[7] = (pulse_position[7] << 1) + ((fixed_index[6] >> 1) & 1);

471

472 fixed_sparse->n = 8;

473 for (i = 0; i < 4; i++) {

474 const int pos1 = (pulse_position[i] << 2) + i;

475 const int pos2 = (pulse_position[i + 4] << 2) + i;

476 const float sign = fixed_index[i] ? -1.0 : 1.0;

477 fixed_sparse->x[i ] = pos1;

478 fixed_sparse->x[i + 4] = pos2;

479 fixed_sparse->y[i ] = sign;

480 fixed_sparse->y[i + 4] = pos2 < pos1 ? -sign : sign;

481 }

482 }

483

484 /**

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

486 * then construct the algebraic codebook vector.

487 *

488 * nb of pulses | bits encoding pulses

489 * For MODE_4k75 or MODE_5k15, 2 | 1-3, 4-6, 7

490 * MODE_5k9, 2 | 1, 2-4, 5-6, 7-9

491 * MODE_6k7, 3 | 1-3, 4, 5-7, 8, 9-11

492 * MODE_7k4 or MODE_7k95, 4 | 1-3, 4-6, 7-9, 10, 11-13

493 *

494 * @param fixed_sparse pointer to the algebraic codebook vector

495 * @param pulses algebraic codebook indexes

496 * @param mode mode of the current frame

497 * @param subframe current subframe number

498 */

499 static void decode_fixed_sparse(AMRFixed *fixed_sparse, const uint16_t *pulses,

500 const enum Mode mode, const int subframe)

501 {

502 av_assert1(MODE_4k75 <= (signed)mode && mode <= MODE_12k2);

503

504 if (mode == MODE_12k2) {

505 ff_decode_10_pulses_35bits(pulses, fixed_sparse, gray_decode, 5, 3);

506 } else if (mode == MODE_10k2) {

507 decode_8_pulses_31bits(pulses, fixed_sparse);

508 } else {

509 int *pulse_position = fixed_sparse->x;

510 int i, pulse_subset;

511 const int fixed_index = pulses[0];

512

513 if (mode <= MODE_5k15) {

514 pulse_subset = ((fixed_index >> 3) & 8) + (subframe << 1);

515 pulse_position[0] = ( fixed_index & 7) * 5 + track_position[pulse_subset];

516 pulse_position[1] = ((fixed_index >> 3) & 7) * 5 + track_position[pulse_subset + 1];

517 fixed_sparse->n = 2;

518 } else if (mode == MODE_5k9) {

519 pulse_subset = ((fixed_index & 1) << 1) + 1;

520 pulse_position[0] = ((fixed_index >> 1) & 7) * 5 + pulse_subset;

521 pulse_subset = (fixed_index >> 4) & 3;

522 pulse_position[1] = ((fixed_index >> 6) & 7) * 5 + pulse_subset + (pulse_subset == 3 ? 1 : 0);

523 fixed_sparse->n = pulse_position[0] == pulse_position[1] ? 1 : 2;

524 } else if (mode == MODE_6k7) {

525 pulse_position[0] = (fixed_index & 7) * 5;

526 pulse_subset = (fixed_index >> 2) & 2;

527 pulse_position[1] = ((fixed_index >> 4) & 7) * 5 + pulse_subset + 1;

528 pulse_subset = (fixed_index >> 6) & 2;

529 pulse_position[2] = ((fixed_index >> 8) & 7) * 5 + pulse_subset + 2;

530 fixed_sparse->n = 3;

531 } else { // mode <= MODE_7k95

532 pulse_position[0] = gray_decode[ fixed_index & 7];

533 pulse_position[1] = gray_decode[(fixed_index >> 3) & 7] + 1;

534 pulse_position[2] = gray_decode[(fixed_index >> 6) & 7] + 2;

535 pulse_subset = (fixed_index >> 9) & 1;

536 pulse_position[3] = gray_decode[(fixed_index >> 10) & 7] + pulse_subset + 3;

537 fixed_sparse->n = 4;

538 }

539 for (i = 0; i < fixed_sparse->n; i++)

540 fixed_sparse->y[i] = (pulses[1] >> i) & 1 ? 1.0 : -1.0;

541 }

542 }

543

544 /**

545 * Apply pitch lag to obtain the sharpened fixed vector (section 6.1.2)

546 *

547 * @param p the context

548 * @param subframe unpacked amr subframe

549 * @param mode mode of the current frame

550 * @param fixed_sparse sparse respresentation of the fixed vector

551 */

552 static void pitch_sharpening(AMRContext *p, int subframe, enum Mode mode,

553 AMRFixed *fixed_sparse)

554 {

555 // The spec suggests the current pitch gain is always used, but in other

556 // modes the pitch and codebook gains are joinly quantized (sec 5.8.2)

557 // so the codebook gain cannot depend on the quantized pitch gain.

558 if (mode == MODE_12k2)

559 p->beta = FFMIN(p->pitch_gain[4], 1.0);

560

561 fixed_sparse->pitch_lag = p->pitch_lag_int;

562 fixed_sparse->pitch_fac = p->beta;

563

564 // Save pitch sharpening factor for the next subframe

565 // MODE_4k75 only updates on the 2nd and 4th subframes - this follows from

566 // the fact that the gains for two subframes are jointly quantized.

567 if (mode != MODE_4k75 || subframe & 1)

568 p->beta = av_clipf(p->pitch_gain[4], 0.0, SHARP_MAX);

569 }

570 /// @}

571

572

573 /// @name AMR gain decoding functions

574 /// @{

575

576 /**

577 * fixed gain smoothing

578 * Note that where the spec specifies the "spectrum in the q domain"

579 * in section 6.1.4, in fact frequencies should be used.

580 *

581 * @param p the context

582 * @param lsf LSFs for the current subframe, in the range [0,1]

583 * @param lsf_avg averaged LSFs

584 * @param mode mode of the current frame

585 *

586 * @return fixed gain smoothed

587 */

588 static float fixed_gain_smooth(AMRContext *p , const float *lsf,

589 const float *lsf_avg, const enum Mode mode)

590 {

591 float diff = 0.0;

592 int i;

593

594 for (i = 0; i < LP_FILTER_ORDER; i++)

595 diff += fabs(lsf_avg[i] - lsf[i]) / lsf_avg[i];

596

597 // If diff is large for ten subframes, disable smoothing for a 40-subframe

598 // hangover period.

599 p->diff_count++;

600 if (diff <= 0.65)

601 p->diff_count = 0;

602

603 if (p->diff_count > 10) {

604 p->hang_count = 0;

605 p->diff_count--; // don't let diff_count overflow

606 }

607

608 if (p->hang_count < 40) {

609 p->hang_count++;

610 } else if (mode < MODE_7k4 || mode == MODE_10k2) {

611 const float smoothing_factor = av_clipf(4.0 * diff - 1.6, 0.0, 1.0);

612 const float fixed_gain_mean = (p->fixed_gain[0] + p->fixed_gain[1] +

613 p->fixed_gain[2] + p->fixed_gain[3] +

614 p->fixed_gain[4]) * 0.2;

615 return smoothing_factor * p->fixed_gain[4] +

616 (1.0 - smoothing_factor) * fixed_gain_mean;

617 }

618 return p->fixed_gain[4];

619 }

620

621 /**

622 * Decode pitch gain and fixed gain factor (part of section 6.1.3).

623 *

624 * @param p the context

625 * @param amr_subframe unpacked amr subframe

626 * @param mode mode of the current frame

627 * @param subframe current subframe number

628 * @param fixed_gain_factor decoded gain correction factor

629 */

630 static void decode_gains(AMRContext *p, const AMRNBSubframe *amr_subframe,

631 const enum Mode mode, const int subframe,

632 float *fixed_gain_factor)

633 {

634 if (mode == MODE_12k2 || mode == MODE_7k95) {

635 p->pitch_gain[4] = qua_gain_pit [amr_subframe->p_gain ]

636 * (1.0 / 16384.0);

637 *fixed_gain_factor = qua_gain_code[amr_subframe->fixed_gain]

638 * (1.0 / 2048.0);

639 } else {

640 const uint16_t *gains;

641

642 if (mode >= MODE_6k7) {

643 gains = gains_high[amr_subframe->p_gain];

644 } else if (mode >= MODE_5k15) {

645 gains = gains_low [amr_subframe->p_gain];

646 } else {

647 // gain index is only coded in subframes 0,2 for MODE_4k75

648 gains = gains_MODE_4k75[(p->frame.subframe[subframe & 2].p_gain << 1) + (subframe & 1)];

649 }

650

651 p->pitch_gain[4] = gains[0] * (1.0 / 16384.0);

652 *fixed_gain_factor = gains[1] * (1.0 / 4096.0);

653 }

654 }

655

656 /// @}

657

658

659 /// @name AMR preprocessing functions

660 /// @{

661

662 /**

663 * Circularly convolve a sparse fixed vector with a phase dispersion impulse

664 * response filter (D.6.2 of G.729 and 6.1.5 of AMR).

665 *

666 * @param out vector with filter applied

667 * @param in source vector

668 * @param filter phase filter coefficients

669 *

670 * out[n] = sum(i,0,len-1){ in[i] * filter[(len + n - i)%len] }

671 */

672 static void apply_ir_filter(float *out, const AMRFixed *in,

673 const float *filter)

674 {

675 float filter1[AMR_SUBFRAME_SIZE], ///< filters at pitch lag*1 and *2

676 filter2[AMR_SUBFRAME_SIZE];

677 int lag = in->pitch_lag;

678 float fac = in->pitch_fac;

679 int i;

680

681 if (lag < AMR_SUBFRAME_SIZE) {

682 ff_celp_circ_addf(filter1, filter, filter, lag, fac,

683 AMR_SUBFRAME_SIZE);

684

685 if (lag < AMR_SUBFRAME_SIZE >> 1)

686 ff_celp_circ_addf(filter2, filter, filter1, lag, fac,

687 AMR_SUBFRAME_SIZE);

688 }

689

690 memset(out, 0, sizeof(float) * AMR_SUBFRAME_SIZE);

691 for (i = 0; i < in->n; i++) {

692 int x = in->x[i];

693 float y = in->y[i];

694 const float *filterp;

695

696 if (x >= AMR_SUBFRAME_SIZE - lag) {

697 filterp = filter;

698 } else if (x >= AMR_SUBFRAME_SIZE - (lag << 1)) {

699 filterp = filter1;

700 } else

701 filterp = filter2;

702

703 ff_celp_circ_addf(out, out, filterp, x, y, AMR_SUBFRAME_SIZE);

704 }

705 }

706

707 /**

708 * Reduce fixed vector sparseness by smoothing with one of three IR filters.

709 * Also know as "adaptive phase dispersion".

710 *

711 * This implements 3GPP TS 26.090 section 6.1(5).

712 *

713 * @param p the context

714 * @param fixed_sparse algebraic codebook vector

715 * @param fixed_vector unfiltered fixed vector

716 * @param fixed_gain smoothed gain

717 * @param out space for modified vector if necessary

718 */

719 static const float *anti_sparseness(AMRContext *p, AMRFixed *fixed_sparse,

720 const float *fixed_vector,

721 float fixed_gain, float *out)

722 {

723 int ir_filter_nr;

724

725 if (p->pitch_gain[4] < 0.6) {

726 ir_filter_nr = 0; // strong filtering

727 } else if (p->pitch_gain[4] < 0.9) {

728 ir_filter_nr = 1; // medium filtering

729 } else

730 ir_filter_nr = 2; // no filtering

731

732 // detect 'onset'

733 if (fixed_gain > 2.0 * p->prev_sparse_fixed_gain) {

734 p->ir_filter_onset = 2;

735 } else if (p->ir_filter_onset)

736 p->ir_filter_onset--;

737

738 if (!p->ir_filter_onset) {

739 int i, count = 0;

740

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

742 if (p->pitch_gain[i] < 0.6)

743 count++;

744 if (count > 2)

745 ir_filter_nr = 0;

746

747 if (ir_filter_nr > p->prev_ir_filter_nr + 1)

748 ir_filter_nr--;

749 } else if (ir_filter_nr < 2)

750 ir_filter_nr++;

751

752 // Disable filtering for very low level of fixed_gain.

753 // Note this step is not specified in the technical description but is in

754 // the reference source in the function Ph_disp.

755 if (fixed_gain < 5.0)

756 ir_filter_nr = 2;

757

758 if (p->cur_frame_mode != MODE_7k4 && p->cur_frame_mode < MODE_10k2

759 && ir_filter_nr < 2) {

760 apply_ir_filter(out, fixed_sparse,

761 (p->cur_frame_mode == MODE_7k95 ?

762 ir_filters_lookup_MODE_7k95 :

763 ir_filters_lookup)[ir_filter_nr]);

764 fixed_vector = out;

765 }

766

767 // update ir filter strength history

768 p->prev_ir_filter_nr = ir_filter_nr;

769 p->prev_sparse_fixed_gain = fixed_gain;

770

771 return fixed_vector;

772 }

773

774 /// @}

775

776

777 /// @name AMR synthesis functions

778 /// @{

779

780 /**

781 * Conduct 10th order linear predictive coding synthesis.

782 *

783 * @param p pointer to the AMRContext

784 * @param lpc pointer to the LPC coefficients

785 * @param fixed_gain fixed codebook gain for synthesis

786 * @param fixed_vector algebraic codebook vector

787 * @param samples pointer to the output speech samples

788 * @param overflow 16-bit overflow flag

789 */

790 static int synthesis(AMRContext *p, float *lpc,

791 float fixed_gain, const float *fixed_vector,

792 float *samples, uint8_t overflow)

793 {

794 int i;

795 float excitation[AMR_SUBFRAME_SIZE];

796

797 // if an overflow has been detected, the pitch vector is scaled down by a

798 // factor of 4

799 if (overflow)

800 for (i = 0; i < AMR_SUBFRAME_SIZE; i++)

801 p->pitch_vector[i] *= 0.25;

802

803 p->acelpv_ctx.weighted_vector_sumf(excitation, p->pitch_vector, fixed_vector,

804 p->pitch_gain[4], fixed_gain, AMR_SUBFRAME_SIZE);

805

806 // emphasize pitch vector contribution

807 if (p->pitch_gain[4] > 0.5 && !overflow) {

808 float energy = p->celpm_ctx.dot_productf(excitation, excitation,

809 AMR_SUBFRAME_SIZE);

810 float pitch_factor =

811 p->pitch_gain[4] *

812 (p->cur_frame_mode == MODE_12k2 ?

813 0.25 * FFMIN(p->pitch_gain[4], 1.0) :

814 0.5 * FFMIN(p->pitch_gain[4], SHARP_MAX));

815

816 for (i = 0; i < AMR_SUBFRAME_SIZE; i++)

817 excitation[i] += pitch_factor * p->pitch_vector[i];

818

819 ff_scale_vector_to_given_sum_of_squares(excitation, excitation, energy,

820 AMR_SUBFRAME_SIZE);

821 }

822

823 p->celpf_ctx.celp_lp_synthesis_filterf(samples, lpc, excitation,

824 AMR_SUBFRAME_SIZE,

825 LP_FILTER_ORDER);

826

827 // detect overflow

828 for (i = 0; i < AMR_SUBFRAME_SIZE; i++)

829 if (fabsf(samples[i]) > AMR_SAMPLE_BOUND) {

830 return 1;

831 }

832

833 return 0;

834 }

835

836 /// @}

837

838

839 /// @name AMR update functions

840 /// @{

841

842 /**

843 * Update buffers and history at the end of decoding a subframe.

844 *

845 * @param p pointer to the AMRContext

846 */

847 static void update_state(AMRContext *p)

848 {

849 memcpy(p->prev_lsp_sub4, p->lsp[3], LP_FILTER_ORDER * sizeof(p->lsp[3][0]));

850

851 memmove(&p->excitation_buf[0], &p->excitation_buf[AMR_SUBFRAME_SIZE],

852 (PITCH_DELAY_MAX + LP_FILTER_ORDER + 1) * sizeof(float));

853

854 memmove(&p->pitch_gain[0], &p->pitch_gain[1], 4 * sizeof(float));

855 memmove(&p->fixed_gain[0], &p->fixed_gain[1], 4 * sizeof(float));

856

857 memmove(&p->samples_in[0], &p->samples_in[AMR_SUBFRAME_SIZE],

858 LP_FILTER_ORDER * sizeof(float));

859 }

860

861 /// @}

862

863

864 /// @name AMR Postprocessing functions

865 /// @{

866

867 /**

868 * Get the tilt factor of a formant filter from its transfer function

869 *

870 * @param p The Context

871 * @param lpc_n LP_FILTER_ORDER coefficients of the numerator

872 * @param lpc_d LP_FILTER_ORDER coefficients of the denominator

873 */

874 static float tilt_factor(AMRContext *p, float *lpc_n, float *lpc_d)

875 {

876 float rh0, rh1; // autocorrelation at lag 0 and 1

877

878 // LP_FILTER_ORDER prior zeros are needed for ff_celp_lp_synthesis_filterf

879 float impulse_buffer[LP_FILTER_ORDER + AMR_TILT_RESPONSE] = { 0 };

880 float *hf = impulse_buffer + LP_FILTER_ORDER; // start of impulse response

881

882 hf[0] = 1.0;

883 memcpy(hf + 1, lpc_n, sizeof(float) * LP_FILTER_ORDER);

884 p->celpf_ctx.celp_lp_synthesis_filterf(hf, lpc_d, hf,

885 AMR_TILT_RESPONSE,

886 LP_FILTER_ORDER);

887

888 rh0 = p->celpm_ctx.dot_productf(hf, hf, AMR_TILT_RESPONSE);

889 rh1 = p->celpm_ctx.dot_productf(hf, hf + 1, AMR_TILT_RESPONSE - 1);

890

891 // The spec only specifies this check for 12.2 and 10.2 kbit/s

892 // modes. But in the ref source the tilt is always non-negative.

893 return rh1 >= 0.0 ? rh1 / rh0 * AMR_TILT_GAMMA_T : 0.0;

894 }

895

896 /**

897 * Perform adaptive post-filtering to enhance the quality of the speech.

898 * See section 6.2.1.

899 *

900 * @param p pointer to the AMRContext

901 * @param lpc interpolated LP coefficients for this subframe

902 * @param buf_out output of the filter

903 */

904 static void postfilter(AMRContext *p, float *lpc, float *buf_out)

905 {

906 int i;

907 float *samples = p->samples_in + LP_FILTER_ORDER; // Start of input

908

909 float speech_gain = p->celpm_ctx.dot_productf(samples, samples,

910 AMR_SUBFRAME_SIZE);

911

912 float pole_out[AMR_SUBFRAME_SIZE + LP_FILTER_ORDER]; // Output of pole filter

913 const float *gamma_n, *gamma_d; // Formant filter factor table

914 float lpc_n[LP_FILTER_ORDER], lpc_d[LP_FILTER_ORDER]; // Transfer function coefficients

915

916 if (p->cur_frame_mode == MODE_12k2 || p->cur_frame_mode == MODE_10k2) {

917 gamma_n = ff_pow_0_7;

918 gamma_d = ff_pow_0_75;

919 } else {

920 gamma_n = ff_pow_0_55;

921 gamma_d = ff_pow_0_7;

922 }

923

924 for (i = 0; i < LP_FILTER_ORDER; i++) {

925 lpc_n[i] = lpc[i] * gamma_n[i];

926 lpc_d[i] = lpc[i] * gamma_d[i];

927 }

928

929 memcpy(pole_out, p->postfilter_mem, sizeof(float) * LP_FILTER_ORDER);

930 p->celpf_ctx.celp_lp_synthesis_filterf(pole_out + LP_FILTER_ORDER, lpc_d, samples,

931 AMR_SUBFRAME_SIZE, LP_FILTER_ORDER);

932 memcpy(p->postfilter_mem, pole_out + AMR_SUBFRAME_SIZE,

933 sizeof(float) * LP_FILTER_ORDER);

934

935 p->celpf_ctx.celp_lp_zero_synthesis_filterf(buf_out, lpc_n,

936 pole_out + LP_FILTER_ORDER,

937 AMR_SUBFRAME_SIZE, LP_FILTER_ORDER);

938

939 ff_tilt_compensation(&p->tilt_mem, tilt_factor(p, lpc_n, lpc_d), buf_out,

940 AMR_SUBFRAME_SIZE);

941

942 ff_adaptive_gain_control(buf_out, buf_out, speech_gain, AMR_SUBFRAME_SIZE,

943 AMR_AGC_ALPHA, &p->postfilter_agc);

944 }

945

946 /// @}

947

948 static int amrnb_decode_frame(AVCodecContext *avctx, void *data,

949 int *got_frame_ptr, AVPacket *avpkt)

950 {

951

952 AMRContext *p = avctx->priv_data; // pointer to private data

953 AVFrame *frame = data;

954 const uint8_t *buf = avpkt->data;

955 int buf_size = avpkt->size;

956 float *buf_out; // pointer to the output data buffer

957 int i, subframe, ret;

958 float fixed_gain_factor;

959 AMRFixed fixed_sparse = {0}; // fixed vector up to anti-sparseness processing

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

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

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

963

964 /* get output buffer */

965 frame->nb_samples = AMR_BLOCK_SIZE;

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

967 return ret;

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

969

970 p->cur_frame_mode = unpack_bitstream(p, buf, buf_size);

971 if (p->cur_frame_mode == NO_DATA) {

972 av_log(avctx, AV_LOG_ERROR, "Corrupt bitstream\n");

973 return AVERROR_INVALIDDATA;

974 }

975 if (p->cur_frame_mode == MODE_DTX) {

976 avpriv_report_missing_feature(avctx, "dtx mode");

977 av_log(avctx, AV_LOG_INFO, "Note: libopencore_amrnb supports dtx\n");

978 return AVERROR_PATCHWELCOME;

979 }

980

981 if (p->cur_frame_mode == MODE_12k2) {

982 lsf2lsp_5(p);

983 } else

984 lsf2lsp_3(p);

985

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

987 ff_acelp_lspd2lpc(p->lsp[i], p->lpc[i], 5);

988

989 for (subframe = 0; subframe < 4; subframe++) {

990 const AMRNBSubframe *amr_subframe = &p->frame.subframe[subframe];

991

992 decode_pitch_vector(p, amr_subframe, subframe);

993

994 decode_fixed_sparse(&fixed_sparse, amr_subframe->pulses,

995 p->cur_frame_mode, subframe);

996

997 // The fixed gain (section 6.1.3) depends on the fixed vector

998 // (section 6.1.2), but the fixed vector calculation uses

999 // pitch sharpening based on the on the pitch gain (section 6.1.3).

1000 // So the correct order is: pitch gain, pitch sharpening, fixed gain.

1001 decode_gains(p, amr_subframe, p->cur_frame_mode, subframe,

1002 &fixed_gain_factor);

1003

1004 pitch_sharpening(p, subframe, p->cur_frame_mode, &fixed_sparse);

1005

1006 if (fixed_sparse.pitch_lag == 0) {

1007 av_log(avctx, AV_LOG_ERROR, "The file is corrupted, pitch_lag = 0 is not allowed\n");

1008 return AVERROR_INVALIDDATA;

1009 }

1010 ff_set_fixed_vector(p->fixed_vector, &fixed_sparse, 1.0,

1011 AMR_SUBFRAME_SIZE);

1012

1013 p->fixed_gain[4] =

1014 ff_amr_set_fixed_gain(fixed_gain_factor,

1015 p->celpm_ctx.dot_productf(p->fixed_vector,

1016 p->fixed_vector,

1017 AMR_SUBFRAME_SIZE) /

1018 AMR_SUBFRAME_SIZE,

1019 p->prediction_error,

1020 energy_mean[p->cur_frame_mode], energy_pred_fac);

1021

1022 // The excitation feedback is calculated without any processing such

1023 // as fixed gain smoothing. This isn't mentioned in the specification.

1024 for (i = 0; i < AMR_SUBFRAME_SIZE; i++)

1025 p->excitation[i] *= p->pitch_gain[4];

1026 ff_set_fixed_vector(p->excitation, &fixed_sparse, p->fixed_gain[4],

1027 AMR_SUBFRAME_SIZE);

1028

1029 // In the ref decoder, excitation is stored with no fractional bits.

1030 // This step prevents buzz in silent periods. The ref encoder can

1031 // emit long sequences with pitch factor greater than one. This

1032 // creates unwanted feedback if the excitation vector is nonzero.

1033 // (e.g. test sequence T19_795.COD in 3GPP TS 26.074)

1034 for (i = 0; i < AMR_SUBFRAME_SIZE; i++)

1035 p->excitation[i] = truncf(p->excitation[i]);

1036

1037 // Smooth fixed gain.

1038 // The specification is ambiguous, but in the reference source, the

1039 // smoothed value is NOT fed back into later fixed gain smoothing.

1040 synth_fixed_gain = fixed_gain_smooth(p, p->lsf_q[subframe],

1041 p->lsf_avg, p->cur_frame_mode);

1042

1043 synth_fixed_vector = anti_sparseness(p, &fixed_sparse, p->fixed_vector,

1044 synth_fixed_gain, spare_vector);

1045

1046 if (synthesis(p, p->lpc[subframe], synth_fixed_gain,

1047 synth_fixed_vector, &p->samples_in[LP_FILTER_ORDER], 0))

1048 // overflow detected -> rerun synthesis scaling pitch vector down

1049 // by a factor of 4, skipping pitch vector contribution emphasis

1050 // and adaptive gain control

1051 synthesis(p, p->lpc[subframe], synth_fixed_gain,

1052 synth_fixed_vector, &p->samples_in[LP_FILTER_ORDER], 1);

1053

1054 postfilter(p, p->lpc[subframe], buf_out + subframe * AMR_SUBFRAME_SIZE);

1055

1056 // update buffers and history

1057 ff_clear_fixed_vector(p->fixed_vector, &fixed_sparse, AMR_SUBFRAME_SIZE);

1058 update_state(p);

1059 }

1060

1061 p->acelpf_ctx.acelp_apply_order_2_transfer_function(buf_out,

1062 buf_out, highpass_zeros,

1063 highpass_poles,

1064 highpass_gain * AMR_SAMPLE_SCALE,

1065 p->high_pass_mem, AMR_BLOCK_SIZE);

1066

1067 /* Update averaged lsf vector (used for fixed gain smoothing).

1068 *

1069 * Note that lsf_avg should not incorporate the current frame's LSFs

1070 * for fixed_gain_smooth.

1071 * The specification has an incorrect formula: the reference decoder uses

1072 * qbar(n-1) rather than qbar(n) in section 6.1(4) equation 71. */

1073 p->acelpv_ctx.weighted_vector_sumf(p->lsf_avg, p->lsf_avg, p->lsf_q[3],

1074 0.84, 0.16, LP_FILTER_ORDER);

1075

1076 *got_frame_ptr = 1;

1077

1078 /* return the amount of bytes consumed if everything was OK */

1079 return frame_sizes_nb[p->cur_frame_mode] + 1; // +7 for rounding and +8 for TOC

1080 }

1081

1082

1083 AVCodec ff_amrnb_decoder = {

1084 .name = "amrnb",

1085 .long_name = NULL_IF_CONFIG_SMALL("AMR-NB (Adaptive Multi-Rate NarrowBand)"),

1086 .type = AVMEDIA_TYPE_AUDIO,

1087 .id = AV_CODEC_ID_AMR_NB,

1088 .priv_data_size = sizeof(AMRContext),

1089 .init = amrnb_decode_init,

1090 .decode = amrnb_decode_frame,

1091 .capabilities = CODEC_CAP_DR1,

1092 .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT,

1093 AV_SAMPLE_FMT_NONE },

1094 };

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