3 * Copyright (c) 2010 Marcelo Galvao Povoa
5 * This file is part of Libav.
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9 * License as published by the Free Software Foundation; either
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17 * You should have received a copy of the GNU Lesser General Public
18 * License along with Libav; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
24 * AMR wideband decoder
27 #include "libavutil/channel_layout.h"
28 #include "libavutil/common.h"
29 #include "libavutil/float_dsp.h"
30 #include "libavutil/lfg.h"
34 #include "celp_filters.h"
35 #include "acelp_filters.h"
36 #include "acelp_vectors.h"
37 #include "acelp_pitch_delay.h"
40 #define AMR_USE_16BIT_TABLES
43 #include "amrwbdata.h"
45 typedef struct AMRWBContext {
46 AMRWBFrame frame; ///< AMRWB parameters decoded from bitstream
47 enum Mode fr_cur_mode; ///< mode index of current frame
48 uint8_t fr_quality; ///< frame quality index (FQI)
49 float isf_cur[LP_ORDER]; ///< working ISF vector from current frame
50 float isf_q_past[LP_ORDER]; ///< quantized ISF vector of the previous frame
51 float isf_past_final[LP_ORDER]; ///< final processed ISF vector of the previous frame
52 double isp[4][LP_ORDER]; ///< ISP vectors from current frame
53 double isp_sub4_past[LP_ORDER]; ///< ISP vector for the 4th subframe of the previous frame
55 float lp_coef[4][LP_ORDER]; ///< Linear Prediction Coefficients from ISP vector
57 uint8_t base_pitch_lag; ///< integer part of pitch lag for the next relative subframe
58 uint8_t pitch_lag_int; ///< integer part of pitch lag of the previous subframe
60 float excitation_buf[AMRWB_P_DELAY_MAX + LP_ORDER + 2 + AMRWB_SFR_SIZE]; ///< current excitation and all necessary excitation history
61 float *excitation; ///< points to current excitation in excitation_buf[]
63 float pitch_vector[AMRWB_SFR_SIZE]; ///< adaptive codebook (pitch) vector for current subframe
64 float fixed_vector[AMRWB_SFR_SIZE]; ///< algebraic codebook (fixed) vector for current subframe
66 float prediction_error[4]; ///< quantified prediction errors {20log10(^gamma_gc)} for previous four subframes
67 float pitch_gain[6]; ///< quantified pitch gains for the current and previous five subframes
68 float fixed_gain[2]; ///< quantified fixed gains for the current and previous subframes
70 float tilt_coef; ///< {beta_1} related to the voicing of the previous subframe
72 float prev_sparse_fixed_gain; ///< previous fixed gain; used by anti-sparseness to determine "onset"
73 uint8_t prev_ir_filter_nr; ///< previous impulse response filter "impNr": 0 - strong, 1 - medium, 2 - none
74 float prev_tr_gain; ///< previous initial gain used by noise enhancer for threshold
76 float samples_az[LP_ORDER + AMRWB_SFR_SIZE]; ///< low-band samples and memory from synthesis at 12.8kHz
77 float samples_up[UPS_MEM_SIZE + AMRWB_SFR_SIZE]; ///< low-band samples and memory processed for upsampling
78 float samples_hb[LP_ORDER_16k + AMRWB_SFR_SIZE_16k]; ///< high-band samples and memory from synthesis at 16kHz
80 float hpf_31_mem[2], hpf_400_mem[2]; ///< previous values in the high pass filters
81 float demph_mem[1]; ///< previous value in the de-emphasis filter
82 float bpf_6_7_mem[HB_FIR_SIZE]; ///< previous values in the high-band band pass filter
83 float lpf_7_mem[HB_FIR_SIZE]; ///< previous values in the high-band low pass filter
85 AVLFG prng; ///< random number generator for white noise excitation
86 uint8_t first_frame; ///< flag active during decoding of the first frame
89 static av_cold int amrwb_decode_init(AVCodecContext *avctx)
91 AMRWBContext *ctx = avctx->priv_data;
94 if (avctx->channels > 1) {
95 avpriv_report_missing_feature(avctx, "multi-channel AMR");
96 return AVERROR_PATCHWELCOME;
100 avctx->channel_layout = AV_CH_LAYOUT_MONO;
101 avctx->sample_rate = 16000;
102 avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
104 av_lfg_init(&ctx->prng, 1);
106 ctx->excitation = &ctx->excitation_buf[AMRWB_P_DELAY_MAX + LP_ORDER + 1];
107 ctx->first_frame = 1;
109 for (i = 0; i < LP_ORDER; i++)
110 ctx->isf_past_final[i] = isf_init[i] * (1.0f / (1 << 15));
112 for (i = 0; i < 4; i++)
113 ctx->prediction_error[i] = MIN_ENERGY;
119 * Decode the frame header in the "MIME/storage" format. This format
120 * is simpler and does not carry the auxiliary frame information.
122 * @param[in] ctx The Context
123 * @param[in] buf Pointer to the input buffer
125 * @return The decoded header length in bytes
127 static int decode_mime_header(AMRWBContext *ctx, const uint8_t *buf)
129 /* Decode frame header (1st octet) */
130 ctx->fr_cur_mode = buf[0] >> 3 & 0x0F;
131 ctx->fr_quality = (buf[0] & 0x4) == 0x4;
137 * Decode quantized ISF vectors using 36-bit indexes (6K60 mode only).
139 * @param[in] ind Array of 5 indexes
140 * @param[out] isf_q Buffer for isf_q[LP_ORDER]
142 static void decode_isf_indices_36b(uint16_t *ind, float *isf_q)
146 for (i = 0; i < 9; i++)
147 isf_q[i] = dico1_isf[ind[0]][i] * (1.0f / (1 << 15));
149 for (i = 0; i < 7; i++)
150 isf_q[i + 9] = dico2_isf[ind[1]][i] * (1.0f / (1 << 15));
152 for (i = 0; i < 5; i++)
153 isf_q[i] += dico21_isf_36b[ind[2]][i] * (1.0f / (1 << 15));
155 for (i = 0; i < 4; i++)
156 isf_q[i + 5] += dico22_isf_36b[ind[3]][i] * (1.0f / (1 << 15));
158 for (i = 0; i < 7; i++)
159 isf_q[i + 9] += dico23_isf_36b[ind[4]][i] * (1.0f / (1 << 15));
163 * Decode quantized ISF vectors using 46-bit indexes (except 6K60 mode).
165 * @param[in] ind Array of 7 indexes
166 * @param[out] isf_q Buffer for isf_q[LP_ORDER]
168 static void decode_isf_indices_46b(uint16_t *ind, float *isf_q)
172 for (i = 0; i < 9; i++)
173 isf_q[i] = dico1_isf[ind[0]][i] * (1.0f / (1 << 15));
175 for (i = 0; i < 7; i++)
176 isf_q[i + 9] = dico2_isf[ind[1]][i] * (1.0f / (1 << 15));
178 for (i = 0; i < 3; i++)
179 isf_q[i] += dico21_isf[ind[2]][i] * (1.0f / (1 << 15));
181 for (i = 0; i < 3; i++)
182 isf_q[i + 3] += dico22_isf[ind[3]][i] * (1.0f / (1 << 15));
184 for (i = 0; i < 3; i++)
185 isf_q[i + 6] += dico23_isf[ind[4]][i] * (1.0f / (1 << 15));
187 for (i = 0; i < 3; i++)
188 isf_q[i + 9] += dico24_isf[ind[5]][i] * (1.0f / (1 << 15));
190 for (i = 0; i < 4; i++)
191 isf_q[i + 12] += dico25_isf[ind[6]][i] * (1.0f / (1 << 15));
195 * Apply mean and past ISF values using the prediction factor.
196 * Updates past ISF vector.
198 * @param[in,out] isf_q Current quantized ISF
199 * @param[in,out] isf_past Past quantized ISF
201 static void isf_add_mean_and_past(float *isf_q, float *isf_past)
206 for (i = 0; i < LP_ORDER; i++) {
208 isf_q[i] += isf_mean[i] * (1.0f / (1 << 15));
209 isf_q[i] += PRED_FACTOR * isf_past[i];
215 * Interpolate the fourth ISP vector from current and past frames
216 * to obtain an ISP vector for each subframe.
218 * @param[in,out] isp_q ISPs for each subframe
219 * @param[in] isp4_past Past ISP for subframe 4
221 static void interpolate_isp(double isp_q[4][LP_ORDER], const double *isp4_past)
225 for (k = 0; k < 3; k++) {
226 float c = isfp_inter[k];
227 for (i = 0; i < LP_ORDER; i++)
228 isp_q[k][i] = (1.0 - c) * isp4_past[i] + c * isp_q[3][i];
233 * Decode an adaptive codebook index into pitch lag (except 6k60, 8k85 modes).
234 * Calculate integer lag and fractional lag always using 1/4 resolution.
235 * In 1st and 3rd subframes the index is relative to last subframe integer lag.
237 * @param[out] lag_int Decoded integer pitch lag
238 * @param[out] lag_frac Decoded fractional pitch lag
239 * @param[in] pitch_index Adaptive codebook pitch index
240 * @param[in,out] base_lag_int Base integer lag used in relative subframes
241 * @param[in] subframe Current subframe index (0 to 3)
243 static void decode_pitch_lag_high(int *lag_int, int *lag_frac, int pitch_index,
244 uint8_t *base_lag_int, int subframe)
246 if (subframe == 0 || subframe == 2) {
247 if (pitch_index < 376) {
248 *lag_int = (pitch_index + 137) >> 2;
249 *lag_frac = pitch_index - (*lag_int << 2) + 136;
250 } else if (pitch_index < 440) {
251 *lag_int = (pitch_index + 257 - 376) >> 1;
252 *lag_frac = (pitch_index - (*lag_int << 1) + 256 - 376) << 1;
253 /* the actual resolution is 1/2 but expressed as 1/4 */
255 *lag_int = pitch_index - 280;
258 /* minimum lag for next subframe */
259 *base_lag_int = av_clip(*lag_int - 8 - (*lag_frac < 0),
260 AMRWB_P_DELAY_MIN, AMRWB_P_DELAY_MAX - 15);
261 // XXX: the spec states clearly that *base_lag_int should be
262 // the nearest integer to *lag_int (minus 8), but the ref code
263 // actually always uses its floor, I'm following the latter
265 *lag_int = (pitch_index + 1) >> 2;
266 *lag_frac = pitch_index - (*lag_int << 2);
267 *lag_int += *base_lag_int;
272 * Decode an adaptive codebook index into pitch lag for 8k85 and 6k60 modes.
273 * The description is analogous to decode_pitch_lag_high, but in 6k60 the
274 * relative index is used for all subframes except the first.
276 static void decode_pitch_lag_low(int *lag_int, int *lag_frac, int pitch_index,
277 uint8_t *base_lag_int, int subframe, enum Mode mode)
279 if (subframe == 0 || (subframe == 2 && mode != MODE_6k60)) {
280 if (pitch_index < 116) {
281 *lag_int = (pitch_index + 69) >> 1;
282 *lag_frac = (pitch_index - (*lag_int << 1) + 68) << 1;
284 *lag_int = pitch_index - 24;
287 // XXX: same problem as before
288 *base_lag_int = av_clip(*lag_int - 8 - (*lag_frac < 0),
289 AMRWB_P_DELAY_MIN, AMRWB_P_DELAY_MAX - 15);
291 *lag_int = (pitch_index + 1) >> 1;
292 *lag_frac = (pitch_index - (*lag_int << 1)) << 1;
293 *lag_int += *base_lag_int;
298 * Find the pitch vector by interpolating the past excitation at the
299 * pitch delay, which is obtained in this function.
301 * @param[in,out] ctx The context
302 * @param[in] amr_subframe Current subframe data
303 * @param[in] subframe Current subframe index (0 to 3)
305 static void decode_pitch_vector(AMRWBContext *ctx,
306 const AMRWBSubFrame *amr_subframe,
309 int pitch_lag_int, pitch_lag_frac;
311 float *exc = ctx->excitation;
312 enum Mode mode = ctx->fr_cur_mode;
314 if (mode <= MODE_8k85) {
315 decode_pitch_lag_low(&pitch_lag_int, &pitch_lag_frac, amr_subframe->adap,
316 &ctx->base_pitch_lag, subframe, mode);
318 decode_pitch_lag_high(&pitch_lag_int, &pitch_lag_frac, amr_subframe->adap,
319 &ctx->base_pitch_lag, subframe);
321 ctx->pitch_lag_int = pitch_lag_int;
322 pitch_lag_int += pitch_lag_frac > 0;
324 /* Calculate the pitch vector by interpolating the past excitation at the
325 pitch lag using a hamming windowed sinc function */
326 ff_acelp_interpolatef(exc, exc + 1 - pitch_lag_int,
328 pitch_lag_frac + (pitch_lag_frac > 0 ? 0 : 4),
329 LP_ORDER, AMRWB_SFR_SIZE + 1);
331 /* Check which pitch signal path should be used
332 * 6k60 and 8k85 modes have the ltp flag set to 0 */
333 if (amr_subframe->ltp) {
334 memcpy(ctx->pitch_vector, exc, AMRWB_SFR_SIZE * sizeof(float));
336 for (i = 0; i < AMRWB_SFR_SIZE; i++)
337 ctx->pitch_vector[i] = 0.18 * exc[i - 1] + 0.64 * exc[i] +
339 memcpy(exc, ctx->pitch_vector, AMRWB_SFR_SIZE * sizeof(float));
343 /** Get x bits in the index interval [lsb,lsb+len-1] inclusive */
344 #define BIT_STR(x,lsb,len) (((x) >> (lsb)) & ((1 << (len)) - 1))
346 /** Get the bit at specified position */
347 #define BIT_POS(x, p) (((x) >> (p)) & 1)
350 * The next six functions decode_[i]p_track decode exactly i pulses
351 * positions and amplitudes (-1 or 1) in a subframe track using
352 * an encoded pulse indexing (TS 26.190 section 5.8.2).
354 * The results are given in out[], in which a negative number means
355 * amplitude -1 and vice versa (i.e., ampl(x) = x / abs(x) ).
357 * @param[out] out Output buffer (writes i elements)
358 * @param[in] code Pulse index (no. of bits varies, see below)
359 * @param[in] m (log2) Number of potential positions
360 * @param[in] off Offset for decoded positions
362 static inline void decode_1p_track(int *out, int code, int m, int off)
364 int pos = BIT_STR(code, 0, m) + off; ///code: m+1 bits
366 out[0] = BIT_POS(code, m) ? -pos : pos;
369 static inline void decode_2p_track(int *out, int code, int m, int off) ///code: 2m+1 bits
371 int pos0 = BIT_STR(code, m, m) + off;
372 int pos1 = BIT_STR(code, 0, m) + off;
374 out[0] = BIT_POS(code, 2*m) ? -pos0 : pos0;
375 out[1] = BIT_POS(code, 2*m) ? -pos1 : pos1;
376 out[1] = pos0 > pos1 ? -out[1] : out[1];
379 static void decode_3p_track(int *out, int code, int m, int off) ///code: 3m+1 bits
381 int half_2p = BIT_POS(code, 2*m - 1) << (m - 1);
383 decode_2p_track(out, BIT_STR(code, 0, 2*m - 1),
384 m - 1, off + half_2p);
385 decode_1p_track(out + 2, BIT_STR(code, 2*m, m + 1), m, off);
388 static void decode_4p_track(int *out, int code, int m, int off) ///code: 4m bits
390 int half_4p, subhalf_2p;
391 int b_offset = 1 << (m - 1);
393 switch (BIT_STR(code, 4*m - 2, 2)) { /* case ID (2 bits) */
394 case 0: /* 0 pulses in A, 4 pulses in B or vice versa */
395 half_4p = BIT_POS(code, 4*m - 3) << (m - 1); // which has 4 pulses
396 subhalf_2p = BIT_POS(code, 2*m - 3) << (m - 2);
398 decode_2p_track(out, BIT_STR(code, 0, 2*m - 3),
399 m - 2, off + half_4p + subhalf_2p);
400 decode_2p_track(out + 2, BIT_STR(code, 2*m - 2, 2*m - 1),
401 m - 1, off + half_4p);
403 case 1: /* 1 pulse in A, 3 pulses in B */
404 decode_1p_track(out, BIT_STR(code, 3*m - 2, m),
406 decode_3p_track(out + 1, BIT_STR(code, 0, 3*m - 2),
407 m - 1, off + b_offset);
409 case 2: /* 2 pulses in each half */
410 decode_2p_track(out, BIT_STR(code, 2*m - 1, 2*m - 1),
412 decode_2p_track(out + 2, BIT_STR(code, 0, 2*m - 1),
413 m - 1, off + b_offset);
415 case 3: /* 3 pulses in A, 1 pulse in B */
416 decode_3p_track(out, BIT_STR(code, m, 3*m - 2),
418 decode_1p_track(out + 3, BIT_STR(code, 0, m),
419 m - 1, off + b_offset);
424 static void decode_5p_track(int *out, int code, int m, int off) ///code: 5m bits
426 int half_3p = BIT_POS(code, 5*m - 1) << (m - 1);
428 decode_3p_track(out, BIT_STR(code, 2*m + 1, 3*m - 2),
429 m - 1, off + half_3p);
431 decode_2p_track(out + 3, BIT_STR(code, 0, 2*m + 1), m, off);
434 static void decode_6p_track(int *out, int code, int m, int off) ///code: 6m-2 bits
436 int b_offset = 1 << (m - 1);
437 /* which half has more pulses in cases 0 to 2 */
438 int half_more = BIT_POS(code, 6*m - 5) << (m - 1);
439 int half_other = b_offset - half_more;
441 switch (BIT_STR(code, 6*m - 4, 2)) { /* case ID (2 bits) */
442 case 0: /* 0 pulses in A, 6 pulses in B or vice versa */
443 decode_1p_track(out, BIT_STR(code, 0, m),
444 m - 1, off + half_more);
445 decode_5p_track(out + 1, BIT_STR(code, m, 5*m - 5),
446 m - 1, off + half_more);
448 case 1: /* 1 pulse in A, 5 pulses in B or vice versa */
449 decode_1p_track(out, BIT_STR(code, 0, m),
450 m - 1, off + half_other);
451 decode_5p_track(out + 1, BIT_STR(code, m, 5*m - 5),
452 m - 1, off + half_more);
454 case 2: /* 2 pulses in A, 4 pulses in B or vice versa */
455 decode_2p_track(out, BIT_STR(code, 0, 2*m - 1),
456 m - 1, off + half_other);
457 decode_4p_track(out + 2, BIT_STR(code, 2*m - 1, 4*m - 4),
458 m - 1, off + half_more);
460 case 3: /* 3 pulses in A, 3 pulses in B */
461 decode_3p_track(out, BIT_STR(code, 3*m - 2, 3*m - 2),
463 decode_3p_track(out + 3, BIT_STR(code, 0, 3*m - 2),
464 m - 1, off + b_offset);
470 * Decode the algebraic codebook index to pulse positions and signs,
471 * then construct the algebraic codebook vector.
473 * @param[out] fixed_vector Buffer for the fixed codebook excitation
474 * @param[in] pulse_hi MSBs part of the pulse index array (higher modes only)
475 * @param[in] pulse_lo LSBs part of the pulse index array
476 * @param[in] mode Mode of the current frame
478 static void decode_fixed_vector(float *fixed_vector, const uint16_t *pulse_hi,
479 const uint16_t *pulse_lo, const enum Mode mode)
481 /* sig_pos stores for each track the decoded pulse position indexes
482 * (1-based) multiplied by its corresponding amplitude (+1 or -1) */
484 int spacing = (mode == MODE_6k60) ? 2 : 4;
489 for (i = 0; i < 2; i++)
490 decode_1p_track(sig_pos[i], pulse_lo[i], 5, 1);
493 for (i = 0; i < 4; i++)
494 decode_1p_track(sig_pos[i], pulse_lo[i], 4, 1);
497 for (i = 0; i < 4; i++)
498 decode_2p_track(sig_pos[i], pulse_lo[i], 4, 1);
501 for (i = 0; i < 2; i++)
502 decode_3p_track(sig_pos[i], pulse_lo[i], 4, 1);
503 for (i = 2; i < 4; i++)
504 decode_2p_track(sig_pos[i], pulse_lo[i], 4, 1);
507 for (i = 0; i < 4; i++)
508 decode_3p_track(sig_pos[i], pulse_lo[i], 4, 1);
511 for (i = 0; i < 4; i++)
512 decode_4p_track(sig_pos[i], (int) pulse_lo[i] +
513 ((int) pulse_hi[i] << 14), 4, 1);
516 for (i = 0; i < 2; i++)
517 decode_5p_track(sig_pos[i], (int) pulse_lo[i] +
518 ((int) pulse_hi[i] << 10), 4, 1);
519 for (i = 2; i < 4; i++)
520 decode_4p_track(sig_pos[i], (int) pulse_lo[i] +
521 ((int) pulse_hi[i] << 14), 4, 1);
525 for (i = 0; i < 4; i++)
526 decode_6p_track(sig_pos[i], (int) pulse_lo[i] +
527 ((int) pulse_hi[i] << 11), 4, 1);
531 memset(fixed_vector, 0, sizeof(float) * AMRWB_SFR_SIZE);
533 for (i = 0; i < 4; i++)
534 for (j = 0; j < pulses_nb_per_mode_tr[mode][i]; j++) {
535 int pos = (FFABS(sig_pos[i][j]) - 1) * spacing + i;
537 fixed_vector[pos] += sig_pos[i][j] < 0 ? -1.0 : 1.0;
542 * Decode pitch gain and fixed gain correction factor.
544 * @param[in] vq_gain Vector-quantized index for gains
545 * @param[in] mode Mode of the current frame
546 * @param[out] fixed_gain_factor Decoded fixed gain correction factor
547 * @param[out] pitch_gain Decoded pitch gain
549 static void decode_gains(const uint8_t vq_gain, const enum Mode mode,
550 float *fixed_gain_factor, float *pitch_gain)
552 const int16_t *gains = (mode <= MODE_8k85 ? qua_gain_6b[vq_gain] :
553 qua_gain_7b[vq_gain]);
555 *pitch_gain = gains[0] * (1.0f / (1 << 14));
556 *fixed_gain_factor = gains[1] * (1.0f / (1 << 11));
560 * Apply pitch sharpening filters to the fixed codebook vector.
562 * @param[in] ctx The context
563 * @param[in,out] fixed_vector Fixed codebook excitation
565 // XXX: Spec states this procedure should be applied when the pitch
566 // lag is less than 64, but this checking seems absent in reference and AMR-NB
567 static void pitch_sharpening(AMRWBContext *ctx, float *fixed_vector)
572 for (i = AMRWB_SFR_SIZE - 1; i != 0; i--)
573 fixed_vector[i] -= fixed_vector[i - 1] * ctx->tilt_coef;
575 /* Periodicity enhancement part */
576 for (i = ctx->pitch_lag_int; i < AMRWB_SFR_SIZE; i++)
577 fixed_vector[i] += fixed_vector[i - ctx->pitch_lag_int] * 0.85;
581 * Calculate the voicing factor (-1.0 = unvoiced to 1.0 = voiced).
583 * @param[in] p_vector, f_vector Pitch and fixed excitation vectors
584 * @param[in] p_gain, f_gain Pitch and fixed gains
586 // XXX: There is something wrong with the precision here! The magnitudes
587 // of the energies are not correct. Please check the reference code carefully
588 static float voice_factor(float *p_vector, float p_gain,
589 float *f_vector, float f_gain)
591 double p_ener = (double) avpriv_scalarproduct_float_c(p_vector, p_vector,
594 double f_ener = (double) avpriv_scalarproduct_float_c(f_vector, f_vector,
598 return (p_ener - f_ener) / (p_ener + f_ener);
602 * Reduce fixed vector sparseness by smoothing with one of three IR filters,
603 * also known as "adaptive phase dispersion".
605 * @param[in] ctx The context
606 * @param[in,out] fixed_vector Unfiltered fixed vector
607 * @param[out] buf Space for modified vector if necessary
609 * @return The potentially overwritten filtered fixed vector address
611 static float *anti_sparseness(AMRWBContext *ctx,
612 float *fixed_vector, float *buf)
616 if (ctx->fr_cur_mode > MODE_8k85) // no filtering in higher modes
619 if (ctx->pitch_gain[0] < 0.6) {
620 ir_filter_nr = 0; // strong filtering
621 } else if (ctx->pitch_gain[0] < 0.9) {
622 ir_filter_nr = 1; // medium filtering
624 ir_filter_nr = 2; // no filtering
627 if (ctx->fixed_gain[0] > 3.0 * ctx->fixed_gain[1]) {
628 if (ir_filter_nr < 2)
633 for (i = 0; i < 6; i++)
634 if (ctx->pitch_gain[i] < 0.6)
640 if (ir_filter_nr > ctx->prev_ir_filter_nr + 1)
644 /* update ir filter strength history */
645 ctx->prev_ir_filter_nr = ir_filter_nr;
647 ir_filter_nr += (ctx->fr_cur_mode == MODE_8k85);
649 if (ir_filter_nr < 2) {
651 const float *coef = ir_filters_lookup[ir_filter_nr];
653 /* Circular convolution code in the reference
654 * decoder was modified to avoid using one
655 * extra array. The filtered vector is given by:
657 * c2(n) = sum(i,0,len-1){ c(i) * coef( (n - i + len) % len ) }
660 memset(buf, 0, sizeof(float) * AMRWB_SFR_SIZE);
661 for (i = 0; i < AMRWB_SFR_SIZE; i++)
663 ff_celp_circ_addf(buf, buf, coef, i, fixed_vector[i],
672 * Calculate a stability factor {teta} based on distance between
673 * current and past isf. A value of 1 shows maximum signal stability.
675 static float stability_factor(const float *isf, const float *isf_past)
680 for (i = 0; i < LP_ORDER - 1; i++)
681 acc += (isf[i] - isf_past[i]) * (isf[i] - isf_past[i]);
683 // XXX: This part is not so clear from the reference code
684 // the result is more accurate changing the "/ 256" to "* 512"
685 return FFMAX(0.0, 1.25 - acc * 0.8 * 512);
689 * Apply a non-linear fixed gain smoothing in order to reduce
690 * fluctuation in the energy of excitation.
692 * @param[in] fixed_gain Unsmoothed fixed gain
693 * @param[in,out] prev_tr_gain Previous threshold gain (updated)
694 * @param[in] voice_fac Frame voicing factor
695 * @param[in] stab_fac Frame stability factor
697 * @return The smoothed gain
699 static float noise_enhancer(float fixed_gain, float *prev_tr_gain,
700 float voice_fac, float stab_fac)
702 float sm_fac = 0.5 * (1 - voice_fac) * stab_fac;
705 // XXX: the following fixed-point constants used to in(de)crement
706 // gain by 1.5dB were taken from the reference code, maybe it could
708 if (fixed_gain < *prev_tr_gain) {
709 g0 = FFMIN(*prev_tr_gain, fixed_gain + fixed_gain *
710 (6226 * (1.0f / (1 << 15)))); // +1.5 dB
712 g0 = FFMAX(*prev_tr_gain, fixed_gain *
713 (27536 * (1.0f / (1 << 15)))); // -1.5 dB
715 *prev_tr_gain = g0; // update next frame threshold
717 return sm_fac * g0 + (1 - sm_fac) * fixed_gain;
721 * Filter the fixed_vector to emphasize the higher frequencies.
723 * @param[in,out] fixed_vector Fixed codebook vector
724 * @param[in] voice_fac Frame voicing factor
726 static void pitch_enhancer(float *fixed_vector, float voice_fac)
729 float cpe = 0.125 * (1 + voice_fac);
730 float last = fixed_vector[0]; // holds c(i - 1)
732 fixed_vector[0] -= cpe * fixed_vector[1];
734 for (i = 1; i < AMRWB_SFR_SIZE - 1; i++) {
735 float cur = fixed_vector[i];
737 fixed_vector[i] -= cpe * (last + fixed_vector[i + 1]);
741 fixed_vector[AMRWB_SFR_SIZE - 1] -= cpe * last;
745 * Conduct 16th order linear predictive coding synthesis from excitation.
747 * @param[in] ctx Pointer to the AMRWBContext
748 * @param[in] lpc Pointer to the LPC coefficients
749 * @param[out] excitation Buffer for synthesis final excitation
750 * @param[in] fixed_gain Fixed codebook gain for synthesis
751 * @param[in] fixed_vector Algebraic codebook vector
752 * @param[in,out] samples Pointer to the output samples and memory
754 static void synthesis(AMRWBContext *ctx, float *lpc, float *excitation,
755 float fixed_gain, const float *fixed_vector,
758 ff_weighted_vector_sumf(excitation, ctx->pitch_vector, fixed_vector,
759 ctx->pitch_gain[0], fixed_gain, AMRWB_SFR_SIZE);
761 /* emphasize pitch vector contribution in low bitrate modes */
762 if (ctx->pitch_gain[0] > 0.5 && ctx->fr_cur_mode <= MODE_8k85) {
764 float energy = avpriv_scalarproduct_float_c(excitation, excitation,
767 // XXX: Weird part in both ref code and spec. A unknown parameter
768 // {beta} seems to be identical to the current pitch gain
769 float pitch_factor = 0.25 * ctx->pitch_gain[0] * ctx->pitch_gain[0];
771 for (i = 0; i < AMRWB_SFR_SIZE; i++)
772 excitation[i] += pitch_factor * ctx->pitch_vector[i];
774 ff_scale_vector_to_given_sum_of_squares(excitation, excitation,
775 energy, AMRWB_SFR_SIZE);
778 ff_celp_lp_synthesis_filterf(samples, lpc, excitation,
779 AMRWB_SFR_SIZE, LP_ORDER);
783 * Apply to synthesis a de-emphasis filter of the form:
784 * H(z) = 1 / (1 - m * z^-1)
786 * @param[out] out Output buffer
787 * @param[in] in Input samples array with in[-1]
788 * @param[in] m Filter coefficient
789 * @param[in,out] mem State from last filtering
791 static void de_emphasis(float *out, float *in, float m, float mem[1])
795 out[0] = in[0] + m * mem[0];
797 for (i = 1; i < AMRWB_SFR_SIZE; i++)
798 out[i] = in[i] + out[i - 1] * m;
800 mem[0] = out[AMRWB_SFR_SIZE - 1];
804 * Upsample a signal by 5/4 ratio (from 12.8kHz to 16kHz) using
805 * a FIR interpolation filter. Uses past data from before *in address.
807 * @param[out] out Buffer for interpolated signal
808 * @param[in] in Current signal data (length 0.8*o_size)
809 * @param[in] o_size Output signal length
811 static void upsample_5_4(float *out, const float *in, int o_size)
813 const float *in0 = in - UPS_FIR_SIZE + 1;
815 int int_part = 0, frac_part;
818 for (j = 0; j < o_size / 5; j++) {
819 out[i] = in[int_part];
823 for (k = 1; k < 5; k++) {
824 out[i] = avpriv_scalarproduct_float_c(in0 + int_part,
825 upsample_fir[4 - frac_part],
835 * Calculate the high-band gain based on encoded index (23k85 mode) or
836 * on the low-band speech signal and the Voice Activity Detection flag.
838 * @param[in] ctx The context
839 * @param[in] synth LB speech synthesis at 12.8k
840 * @param[in] hb_idx Gain index for mode 23k85 only
841 * @param[in] vad VAD flag for the frame
843 static float find_hb_gain(AMRWBContext *ctx, const float *synth,
844 uint16_t hb_idx, uint8_t vad)
849 if (ctx->fr_cur_mode == MODE_23k85)
850 return qua_hb_gain[hb_idx] * (1.0f / (1 << 14));
852 tilt = avpriv_scalarproduct_float_c(synth, synth + 1, AMRWB_SFR_SIZE - 1) /
853 avpriv_scalarproduct_float_c(synth, synth, AMRWB_SFR_SIZE);
855 /* return gain bounded by [0.1, 1.0] */
856 return av_clipf((1.0 - FFMAX(0.0, tilt)) * (1.25 - 0.25 * wsp), 0.1, 1.0);
860 * Generate the high-band excitation with the same energy from the lower
861 * one and scaled by the given gain.
863 * @param[in] ctx The context
864 * @param[out] hb_exc Buffer for the excitation
865 * @param[in] synth_exc Low-band excitation used for synthesis
866 * @param[in] hb_gain Wanted excitation gain
868 static void scaled_hb_excitation(AMRWBContext *ctx, float *hb_exc,
869 const float *synth_exc, float hb_gain)
872 float energy = avpriv_scalarproduct_float_c(synth_exc, synth_exc,
875 /* Generate a white-noise excitation */
876 for (i = 0; i < AMRWB_SFR_SIZE_16k; i++)
877 hb_exc[i] = 32768.0 - (uint16_t) av_lfg_get(&ctx->prng);
879 ff_scale_vector_to_given_sum_of_squares(hb_exc, hb_exc,
880 energy * hb_gain * hb_gain,
885 * Calculate the auto-correlation for the ISF difference vector.
887 static float auto_correlation(float *diff_isf, float mean, int lag)
892 for (i = 7; i < LP_ORDER - 2; i++) {
893 float prod = (diff_isf[i] - mean) * (diff_isf[i - lag] - mean);
900 * Extrapolate a ISF vector to the 16kHz range (20th order LP)
901 * used at mode 6k60 LP filter for the high frequency band.
903 * @param[out] isf Buffer for extrapolated isf; contains LP_ORDER
906 static void extrapolate_isf(float isf[LP_ORDER_16k])
908 float diff_isf[LP_ORDER - 2], diff_mean;
911 int i, j, i_max_corr;
913 isf[LP_ORDER_16k - 1] = isf[LP_ORDER - 1];
915 /* Calculate the difference vector */
916 for (i = 0; i < LP_ORDER - 2; i++)
917 diff_isf[i] = isf[i + 1] - isf[i];
920 for (i = 2; i < LP_ORDER - 2; i++)
921 diff_mean += diff_isf[i] * (1.0f / (LP_ORDER - 4));
923 /* Find which is the maximum autocorrelation */
925 for (i = 0; i < 3; i++) {
926 corr_lag[i] = auto_correlation(diff_isf, diff_mean, i + 2);
928 if (corr_lag[i] > corr_lag[i_max_corr])
933 for (i = LP_ORDER - 1; i < LP_ORDER_16k - 1; i++)
934 isf[i] = isf[i - 1] + isf[i - 1 - i_max_corr]
935 - isf[i - 2 - i_max_corr];
937 /* Calculate an estimate for ISF(18) and scale ISF based on the error */
938 est = 7965 + (isf[2] - isf[3] - isf[4]) / 6.0;
939 scale = 0.5 * (FFMIN(est, 7600) - isf[LP_ORDER - 2]) /
940 (isf[LP_ORDER_16k - 2] - isf[LP_ORDER - 2]);
942 for (i = LP_ORDER - 1, j = 0; i < LP_ORDER_16k - 1; i++, j++)
943 diff_isf[j] = scale * (isf[i] - isf[i - 1]);
945 /* Stability insurance */
946 for (i = 1; i < LP_ORDER_16k - LP_ORDER; i++)
947 if (diff_isf[i] + diff_isf[i - 1] < 5.0) {
948 if (diff_isf[i] > diff_isf[i - 1]) {
949 diff_isf[i - 1] = 5.0 - diff_isf[i];
951 diff_isf[i] = 5.0 - diff_isf[i - 1];
954 for (i = LP_ORDER - 1, j = 0; i < LP_ORDER_16k - 1; i++, j++)
955 isf[i] = isf[i - 1] + diff_isf[j] * (1.0f / (1 << 15));
957 /* Scale the ISF vector for 16000 Hz */
958 for (i = 0; i < LP_ORDER_16k - 1; i++)
963 * Spectral expand the LP coefficients using the equation:
964 * y[i] = x[i] * (gamma ** i)
966 * @param[out] out Output buffer (may use input array)
967 * @param[in] lpc LP coefficients array
968 * @param[in] gamma Weighting factor
969 * @param[in] size LP array size
971 static void lpc_weighting(float *out, const float *lpc, float gamma, int size)
976 for (i = 0; i < size; i++) {
977 out[i] = lpc[i] * fac;
983 * Conduct 20th order linear predictive coding synthesis for the high
984 * frequency band excitation at 16kHz.
986 * @param[in] ctx The context
987 * @param[in] subframe Current subframe index (0 to 3)
988 * @param[in,out] samples Pointer to the output speech samples
989 * @param[in] exc Generated white-noise scaled excitation
990 * @param[in] isf Current frame isf vector
991 * @param[in] isf_past Past frame final isf vector
993 static void hb_synthesis(AMRWBContext *ctx, int subframe, float *samples,
994 const float *exc, const float *isf, const float *isf_past)
996 float hb_lpc[LP_ORDER_16k];
997 enum Mode mode = ctx->fr_cur_mode;
999 if (mode == MODE_6k60) {
1000 float e_isf[LP_ORDER_16k]; // ISF vector for extrapolation
1001 double e_isp[LP_ORDER_16k];
1003 ff_weighted_vector_sumf(e_isf, isf_past, isf, isfp_inter[subframe],
1004 1.0 - isfp_inter[subframe], LP_ORDER);
1006 extrapolate_isf(e_isf);
1008 e_isf[LP_ORDER_16k - 1] *= 2.0;
1009 ff_acelp_lsf2lspd(e_isp, e_isf, LP_ORDER_16k);
1010 ff_amrwb_lsp2lpc(e_isp, hb_lpc, LP_ORDER_16k);
1012 lpc_weighting(hb_lpc, hb_lpc, 0.9, LP_ORDER_16k);
1014 lpc_weighting(hb_lpc, ctx->lp_coef[subframe], 0.6, LP_ORDER);
1017 ff_celp_lp_synthesis_filterf(samples, hb_lpc, exc, AMRWB_SFR_SIZE_16k,
1018 (mode == MODE_6k60) ? LP_ORDER_16k : LP_ORDER);
1022 * Apply a 15th order filter to high-band samples.
1023 * The filter characteristic depends on the given coefficients.
1025 * @param[out] out Buffer for filtered output
1026 * @param[in] fir_coef Filter coefficients
1027 * @param[in,out] mem State from last filtering (updated)
1028 * @param[in] in Input speech data (high-band)
1030 * @remark It is safe to pass the same array in in and out parameters
1032 static void hb_fir_filter(float *out, const float fir_coef[HB_FIR_SIZE + 1],
1033 float mem[HB_FIR_SIZE], const float *in)
1036 float data[AMRWB_SFR_SIZE_16k + HB_FIR_SIZE]; // past and current samples
1038 memcpy(data, mem, HB_FIR_SIZE * sizeof(float));
1039 memcpy(data + HB_FIR_SIZE, in, AMRWB_SFR_SIZE_16k * sizeof(float));
1041 for (i = 0; i < AMRWB_SFR_SIZE_16k; i++) {
1043 for (j = 0; j <= HB_FIR_SIZE; j++)
1044 out[i] += data[i + j] * fir_coef[j];
1047 memcpy(mem, data + AMRWB_SFR_SIZE_16k, HB_FIR_SIZE * sizeof(float));
1051 * Update context state before the next subframe.
1053 static void update_sub_state(AMRWBContext *ctx)
1055 memmove(&ctx->excitation_buf[0], &ctx->excitation_buf[AMRWB_SFR_SIZE],
1056 (AMRWB_P_DELAY_MAX + LP_ORDER + 1) * sizeof(float));
1058 memmove(&ctx->pitch_gain[1], &ctx->pitch_gain[0], 5 * sizeof(float));
1059 memmove(&ctx->fixed_gain[1], &ctx->fixed_gain[0], 1 * sizeof(float));
1061 memmove(&ctx->samples_az[0], &ctx->samples_az[AMRWB_SFR_SIZE],
1062 LP_ORDER * sizeof(float));
1063 memmove(&ctx->samples_up[0], &ctx->samples_up[AMRWB_SFR_SIZE],
1064 UPS_MEM_SIZE * sizeof(float));
1065 memmove(&ctx->samples_hb[0], &ctx->samples_hb[AMRWB_SFR_SIZE_16k],
1066 LP_ORDER_16k * sizeof(float));
1069 static int amrwb_decode_frame(AVCodecContext *avctx, void *data,
1070 int *got_frame_ptr, AVPacket *avpkt)
1072 AMRWBContext *ctx = avctx->priv_data;
1073 AVFrame *frame = data;
1074 AMRWBFrame *cf = &ctx->frame;
1075 const uint8_t *buf = avpkt->data;
1076 int buf_size = avpkt->size;
1077 int expected_fr_size, header_size;
1079 float spare_vector[AMRWB_SFR_SIZE]; // extra stack space to hold result from anti-sparseness processing
1080 float fixed_gain_factor; // fixed gain correction factor (gamma)
1081 float *synth_fixed_vector; // pointer to the fixed vector that synthesis should use
1082 float synth_fixed_gain; // the fixed gain that synthesis should use
1083 float voice_fac, stab_fac; // parameters used for gain smoothing
1084 float synth_exc[AMRWB_SFR_SIZE]; // post-processed excitation for synthesis
1085 float hb_exc[AMRWB_SFR_SIZE_16k]; // excitation for the high frequency band
1086 float hb_samples[AMRWB_SFR_SIZE_16k]; // filtered high-band samples from synthesis
1090 /* get output buffer */
1091 frame->nb_samples = 4 * AMRWB_SFR_SIZE_16k;
1092 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
1093 av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
1096 buf_out = (float *)frame->data[0];
1098 header_size = decode_mime_header(ctx, buf);
1099 if (ctx->fr_cur_mode > MODE_SID) {
1100 av_log(avctx, AV_LOG_ERROR,
1101 "Invalid mode %d\n", ctx->fr_cur_mode);
1102 return AVERROR_INVALIDDATA;
1104 expected_fr_size = ((cf_sizes_wb[ctx->fr_cur_mode] + 7) >> 3) + 1;
1106 if (buf_size < expected_fr_size) {
1107 av_log(avctx, AV_LOG_ERROR,
1108 "Frame too small (%d bytes). Truncated file?\n", buf_size);
1110 return AVERROR_INVALIDDATA;
1113 if (!ctx->fr_quality || ctx->fr_cur_mode > MODE_SID)
1114 av_log(avctx, AV_LOG_ERROR, "Encountered a bad or corrupted frame\n");
1116 if (ctx->fr_cur_mode == MODE_SID) { /* Comfort noise frame */
1117 avpriv_request_sample(avctx, "SID mode");
1118 return AVERROR_PATCHWELCOME;
1121 ff_amr_bit_reorder((uint16_t *) &ctx->frame, sizeof(AMRWBFrame),
1122 buf + header_size, amr_bit_orderings_by_mode[ctx->fr_cur_mode]);
1124 /* Decode the quantized ISF vector */
1125 if (ctx->fr_cur_mode == MODE_6k60) {
1126 decode_isf_indices_36b(cf->isp_id, ctx->isf_cur);
1128 decode_isf_indices_46b(cf->isp_id, ctx->isf_cur);
1131 isf_add_mean_and_past(ctx->isf_cur, ctx->isf_q_past);
1132 ff_set_min_dist_lsf(ctx->isf_cur, MIN_ISF_SPACING, LP_ORDER - 1);
1134 stab_fac = stability_factor(ctx->isf_cur, ctx->isf_past_final);
1136 ctx->isf_cur[LP_ORDER - 1] *= 2.0;
1137 ff_acelp_lsf2lspd(ctx->isp[3], ctx->isf_cur, LP_ORDER);
1139 /* Generate a ISP vector for each subframe */
1140 if (ctx->first_frame) {
1141 ctx->first_frame = 0;
1142 memcpy(ctx->isp_sub4_past, ctx->isp[3], LP_ORDER * sizeof(double));
1144 interpolate_isp(ctx->isp, ctx->isp_sub4_past);
1146 for (sub = 0; sub < 4; sub++)
1147 ff_amrwb_lsp2lpc(ctx->isp[sub], ctx->lp_coef[sub], LP_ORDER);
1149 for (sub = 0; sub < 4; sub++) {
1150 const AMRWBSubFrame *cur_subframe = &cf->subframe[sub];
1151 float *sub_buf = buf_out + sub * AMRWB_SFR_SIZE_16k;
1153 /* Decode adaptive codebook (pitch vector) */
1154 decode_pitch_vector(ctx, cur_subframe, sub);
1155 /* Decode innovative codebook (fixed vector) */
1156 decode_fixed_vector(ctx->fixed_vector, cur_subframe->pul_ih,
1157 cur_subframe->pul_il, ctx->fr_cur_mode);
1159 pitch_sharpening(ctx, ctx->fixed_vector);
1161 decode_gains(cur_subframe->vq_gain, ctx->fr_cur_mode,
1162 &fixed_gain_factor, &ctx->pitch_gain[0]);
1164 ctx->fixed_gain[0] =
1165 ff_amr_set_fixed_gain(fixed_gain_factor,
1166 avpriv_scalarproduct_float_c(ctx->fixed_vector,
1170 ctx->prediction_error,
1171 ENERGY_MEAN, energy_pred_fac);
1173 /* Calculate voice factor and store tilt for next subframe */
1174 voice_fac = voice_factor(ctx->pitch_vector, ctx->pitch_gain[0],
1175 ctx->fixed_vector, ctx->fixed_gain[0]);
1176 ctx->tilt_coef = voice_fac * 0.25 + 0.25;
1178 /* Construct current excitation */
1179 for (i = 0; i < AMRWB_SFR_SIZE; i++) {
1180 ctx->excitation[i] *= ctx->pitch_gain[0];
1181 ctx->excitation[i] += ctx->fixed_gain[0] * ctx->fixed_vector[i];
1182 ctx->excitation[i] = truncf(ctx->excitation[i]);
1185 /* Post-processing of excitation elements */
1186 synth_fixed_gain = noise_enhancer(ctx->fixed_gain[0], &ctx->prev_tr_gain,
1187 voice_fac, stab_fac);
1189 synth_fixed_vector = anti_sparseness(ctx, ctx->fixed_vector,
1192 pitch_enhancer(synth_fixed_vector, voice_fac);
1194 synthesis(ctx, ctx->lp_coef[sub], synth_exc, synth_fixed_gain,
1195 synth_fixed_vector, &ctx->samples_az[LP_ORDER]);
1197 /* Synthesis speech post-processing */
1198 de_emphasis(&ctx->samples_up[UPS_MEM_SIZE],
1199 &ctx->samples_az[LP_ORDER], PREEMPH_FAC, ctx->demph_mem);
1201 ff_acelp_apply_order_2_transfer_function(&ctx->samples_up[UPS_MEM_SIZE],
1202 &ctx->samples_up[UPS_MEM_SIZE], hpf_zeros, hpf_31_poles,
1203 hpf_31_gain, ctx->hpf_31_mem, AMRWB_SFR_SIZE);
1205 upsample_5_4(sub_buf, &ctx->samples_up[UPS_FIR_SIZE],
1206 AMRWB_SFR_SIZE_16k);
1208 /* High frequency band (6.4 - 7.0 kHz) generation part */
1209 ff_acelp_apply_order_2_transfer_function(hb_samples,
1210 &ctx->samples_up[UPS_MEM_SIZE], hpf_zeros, hpf_400_poles,
1211 hpf_400_gain, ctx->hpf_400_mem, AMRWB_SFR_SIZE);
1213 hb_gain = find_hb_gain(ctx, hb_samples,
1214 cur_subframe->hb_gain, cf->vad);
1216 scaled_hb_excitation(ctx, hb_exc, synth_exc, hb_gain);
1218 hb_synthesis(ctx, sub, &ctx->samples_hb[LP_ORDER_16k],
1219 hb_exc, ctx->isf_cur, ctx->isf_past_final);
1221 /* High-band post-processing filters */
1222 hb_fir_filter(hb_samples, bpf_6_7_coef, ctx->bpf_6_7_mem,
1223 &ctx->samples_hb[LP_ORDER_16k]);
1225 if (ctx->fr_cur_mode == MODE_23k85)
1226 hb_fir_filter(hb_samples, lpf_7_coef, ctx->lpf_7_mem,
1229 /* Add the low and high frequency bands */
1230 for (i = 0; i < AMRWB_SFR_SIZE_16k; i++)
1231 sub_buf[i] = (sub_buf[i] + hb_samples[i]) * (1.0f / (1 << 15));
1233 /* Update buffers and history */
1234 update_sub_state(ctx);
1237 /* update state for next frame */
1238 memcpy(ctx->isp_sub4_past, ctx->isp[3], LP_ORDER * sizeof(ctx->isp[3][0]));
1239 memcpy(ctx->isf_past_final, ctx->isf_cur, LP_ORDER * sizeof(float));
1243 return expected_fr_size;
1246 AVCodec ff_amrwb_decoder = {
1248 .long_name = NULL_IF_CONFIG_SMALL("AMR-WB (Adaptive Multi-Rate WideBand)"),
1249 .type = AVMEDIA_TYPE_AUDIO,
1250 .id = AV_CODEC_ID_AMR_WB,
1251 .priv_data_size = sizeof(AMRWBContext),
1252 .init = amrwb_decode_init,
1253 .decode = amrwb_decode_frame,
1254 .capabilities = AV_CODEC_CAP_DR1,
1255 .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT,
1256 AV_SAMPLE_FMT_NONE },