3 * Copyright (c) 2010 Marcelo Galvao Povoa
5 * This file is part of FFmpeg.
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.
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.
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
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 "celp_math.h"
36 #include "acelp_filters.h"
37 #include "acelp_vectors.h"
38 #include "acelp_pitch_delay.h"
39 #include "codec_internal.h"
42 #define AMR_USE_16BIT_TABLES
45 #include "amrwbdata.h"
46 #include "mips/amrwbdec_mips.h"
48 typedef struct AMRWBContext {
49 AMRWBFrame frame; ///< AMRWB parameters decoded from bitstream
50 enum Mode fr_cur_mode; ///< mode index of current frame
51 uint8_t fr_quality; ///< frame quality index (FQI)
52 float isf_cur[LP_ORDER]; ///< working ISF vector from current frame
53 float isf_q_past[LP_ORDER]; ///< quantized ISF vector of the previous frame
54 float isf_past_final[LP_ORDER]; ///< final processed ISF vector of the previous frame
55 double isp[4][LP_ORDER]; ///< ISP vectors from current frame
56 double isp_sub4_past[LP_ORDER]; ///< ISP vector for the 4th subframe of the previous frame
58 float lp_coef[4][LP_ORDER]; ///< Linear Prediction Coefficients from ISP vector
60 uint8_t base_pitch_lag; ///< integer part of pitch lag for the next relative subframe
61 uint8_t pitch_lag_int; ///< integer part of pitch lag of the previous subframe
63 float excitation_buf[AMRWB_P_DELAY_MAX + LP_ORDER + 2 + AMRWB_SFR_SIZE]; ///< current excitation and all necessary excitation history
64 float *excitation; ///< points to current excitation in excitation_buf[]
66 float pitch_vector[AMRWB_SFR_SIZE]; ///< adaptive codebook (pitch) vector for current subframe
67 float fixed_vector[AMRWB_SFR_SIZE]; ///< algebraic codebook (fixed) vector for current subframe
69 float prediction_error[4]; ///< quantified prediction errors {20log10(^gamma_gc)} for previous four subframes
70 float pitch_gain[6]; ///< quantified pitch gains for the current and previous five subframes
71 float fixed_gain[2]; ///< quantified fixed gains for the current and previous subframes
73 float tilt_coef; ///< {beta_1} related to the voicing of the previous subframe
75 float prev_sparse_fixed_gain; ///< previous fixed gain; used by anti-sparseness to determine "onset"
76 uint8_t prev_ir_filter_nr; ///< previous impulse response filter "impNr": 0 - strong, 1 - medium, 2 - none
77 float prev_tr_gain; ///< previous initial gain used by noise enhancer for threshold
79 float samples_az[LP_ORDER + AMRWB_SFR_SIZE]; ///< low-band samples and memory from synthesis at 12.8kHz
80 float samples_up[UPS_MEM_SIZE + AMRWB_SFR_SIZE]; ///< low-band samples and memory processed for upsampling
81 float samples_hb[LP_ORDER_16k + AMRWB_SFR_SIZE_16k]; ///< high-band samples and memory from synthesis at 16kHz
83 float hpf_31_mem[2], hpf_400_mem[2]; ///< previous values in the high pass filters
84 float demph_mem[1]; ///< previous value in the de-emphasis filter
85 float bpf_6_7_mem[HB_FIR_SIZE]; ///< previous values in the high-band band pass filter
86 float lpf_7_mem[HB_FIR_SIZE]; ///< previous values in the high-band low pass filter
88 AVLFG prng; ///< random number generator for white noise excitation
89 uint8_t first_frame; ///< flag active during decoding of the first frame
90 ACELPFContext acelpf_ctx; ///< context for filters for ACELP-based codecs
91 ACELPVContext acelpv_ctx; ///< context for vector operations for ACELP-based codecs
92 CELPFContext celpf_ctx; ///< context for filters for CELP-based codecs
93 CELPMContext celpm_ctx; ///< context for fixed point math operations
97 typedef struct AMRWBChannelsContext {
99 } AMRWBChannelsContext;
101 static av_cold int amrwb_decode_init(AVCodecContext *avctx)
103 AMRWBChannelsContext *s = avctx->priv_data;
106 if (avctx->ch_layout.nb_channels > 2) {
107 avpriv_report_missing_feature(avctx, ">2 channel AMR");
108 return AVERROR_PATCHWELCOME;
111 if (!avctx->ch_layout.nb_channels) {
112 av_channel_layout_uninit(&avctx->ch_layout);
113 avctx->ch_layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_MONO;
115 if (!avctx->sample_rate)
116 avctx->sample_rate = 16000;
117 avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
119 for (int ch = 0; ch < avctx->ch_layout.nb_channels; ch++) {
120 AMRWBContext *ctx = &s->ch[ch];
122 av_lfg_init(&ctx->prng, 1);
124 ctx->excitation = &ctx->excitation_buf[AMRWB_P_DELAY_MAX + LP_ORDER + 1];
125 ctx->first_frame = 1;
127 for (i = 0; i < LP_ORDER; i++)
128 ctx->isf_past_final[i] = isf_init[i] * (1.0f / (1 << 15));
130 for (i = 0; i < 4; i++)
131 ctx->prediction_error[i] = MIN_ENERGY;
133 ff_acelp_filter_init(&ctx->acelpf_ctx);
134 ff_acelp_vectors_init(&ctx->acelpv_ctx);
135 ff_celp_filter_init(&ctx->celpf_ctx);
136 ff_celp_math_init(&ctx->celpm_ctx);
143 * Decode the frame header in the "MIME/storage" format. This format
144 * is simpler and does not carry the auxiliary frame information.
146 * @param[in] ctx The Context
147 * @param[in] buf Pointer to the input buffer
149 * @return The decoded header length in bytes
151 static int decode_mime_header(AMRWBContext *ctx, const uint8_t *buf)
153 /* Decode frame header (1st octet) */
154 ctx->fr_cur_mode = buf[0] >> 3 & 0x0F;
155 ctx->fr_quality = (buf[0] & 0x4) == 0x4;
161 * Decode quantized ISF vectors using 36-bit indexes (6K60 mode only).
163 * @param[in] ind Array of 5 indexes
164 * @param[out] isf_q Buffer for isf_q[LP_ORDER]
166 static void decode_isf_indices_36b(uint16_t *ind, float *isf_q)
170 for (i = 0; i < 9; i++)
171 isf_q[i] = dico1_isf[ind[0]][i] * (1.0f / (1 << 15));
173 for (i = 0; i < 7; i++)
174 isf_q[i + 9] = dico2_isf[ind[1]][i] * (1.0f / (1 << 15));
176 for (i = 0; i < 5; i++)
177 isf_q[i] += dico21_isf_36b[ind[2]][i] * (1.0f / (1 << 15));
179 for (i = 0; i < 4; i++)
180 isf_q[i + 5] += dico22_isf_36b[ind[3]][i] * (1.0f / (1 << 15));
182 for (i = 0; i < 7; i++)
183 isf_q[i + 9] += dico23_isf_36b[ind[4]][i] * (1.0f / (1 << 15));
187 * Decode quantized ISF vectors using 46-bit indexes (except 6K60 mode).
189 * @param[in] ind Array of 7 indexes
190 * @param[out] isf_q Buffer for isf_q[LP_ORDER]
192 static void decode_isf_indices_46b(uint16_t *ind, float *isf_q)
196 for (i = 0; i < 9; i++)
197 isf_q[i] = dico1_isf[ind[0]][i] * (1.0f / (1 << 15));
199 for (i = 0; i < 7; i++)
200 isf_q[i + 9] = dico2_isf[ind[1]][i] * (1.0f / (1 << 15));
202 for (i = 0; i < 3; i++)
203 isf_q[i] += dico21_isf[ind[2]][i] * (1.0f / (1 << 15));
205 for (i = 0; i < 3; i++)
206 isf_q[i + 3] += dico22_isf[ind[3]][i] * (1.0f / (1 << 15));
208 for (i = 0; i < 3; i++)
209 isf_q[i + 6] += dico23_isf[ind[4]][i] * (1.0f / (1 << 15));
211 for (i = 0; i < 3; i++)
212 isf_q[i + 9] += dico24_isf[ind[5]][i] * (1.0f / (1 << 15));
214 for (i = 0; i < 4; i++)
215 isf_q[i + 12] += dico25_isf[ind[6]][i] * (1.0f / (1 << 15));
219 * Apply mean and past ISF values using the prediction factor.
220 * Updates past ISF vector.
222 * @param[in,out] isf_q Current quantized ISF
223 * @param[in,out] isf_past Past quantized ISF
225 static void isf_add_mean_and_past(float *isf_q, float *isf_past)
230 for (i = 0; i < LP_ORDER; i++) {
232 isf_q[i] += isf_mean[i] * (1.0f / (1 << 15));
233 isf_q[i] += PRED_FACTOR * isf_past[i];
239 * Interpolate the fourth ISP vector from current and past frames
240 * to obtain an ISP vector for each subframe.
242 * @param[in,out] isp_q ISPs for each subframe
243 * @param[in] isp4_past Past ISP for subframe 4
245 static void interpolate_isp(double isp_q[4][LP_ORDER], const double *isp4_past)
249 for (k = 0; k < 3; k++) {
250 float c = isfp_inter[k];
251 for (i = 0; i < LP_ORDER; i++)
252 isp_q[k][i] = (1.0 - c) * isp4_past[i] + c * isp_q[3][i];
257 * Decode an adaptive codebook index into pitch lag (except 6k60, 8k85 modes).
258 * Calculate integer lag and fractional lag always using 1/4 resolution.
259 * In 1st and 3rd subframes the index is relative to last subframe integer lag.
261 * @param[out] lag_int Decoded integer pitch lag
262 * @param[out] lag_frac Decoded fractional pitch lag
263 * @param[in] pitch_index Adaptive codebook pitch index
264 * @param[in,out] base_lag_int Base integer lag used in relative subframes
265 * @param[in] subframe Current subframe index (0 to 3)
267 static void decode_pitch_lag_high(int *lag_int, int *lag_frac, int pitch_index,
268 uint8_t *base_lag_int, int subframe)
270 if (subframe == 0 || subframe == 2) {
271 if (pitch_index < 376) {
272 *lag_int = (pitch_index + 137) >> 2;
273 *lag_frac = pitch_index - (*lag_int << 2) + 136;
274 } else if (pitch_index < 440) {
275 *lag_int = (pitch_index + 257 - 376) >> 1;
276 *lag_frac = (pitch_index - (*lag_int << 1) + 256 - 376) * 2;
277 /* the actual resolution is 1/2 but expressed as 1/4 */
279 *lag_int = pitch_index - 280;
282 /* minimum lag for next subframe */
283 *base_lag_int = av_clip(*lag_int - 8 - (*lag_frac < 0),
284 AMRWB_P_DELAY_MIN, AMRWB_P_DELAY_MAX - 15);
285 // XXX: the spec states clearly that *base_lag_int should be
286 // the nearest integer to *lag_int (minus 8), but the ref code
287 // actually always uses its floor, I'm following the latter
289 *lag_int = (pitch_index + 1) >> 2;
290 *lag_frac = pitch_index - (*lag_int << 2);
291 *lag_int += *base_lag_int;
296 * Decode an adaptive codebook index into pitch lag for 8k85 and 6k60 modes.
297 * The description is analogous to decode_pitch_lag_high, but in 6k60 the
298 * relative index is used for all subframes except the first.
300 static void decode_pitch_lag_low(int *lag_int, int *lag_frac, int pitch_index,
301 uint8_t *base_lag_int, int subframe, enum Mode mode)
303 if (subframe == 0 || (subframe == 2 && mode != MODE_6k60)) {
304 if (pitch_index < 116) {
305 *lag_int = (pitch_index + 69) >> 1;
306 *lag_frac = (pitch_index - (*lag_int << 1) + 68) * 2;
308 *lag_int = pitch_index - 24;
311 // XXX: same problem as before
312 *base_lag_int = av_clip(*lag_int - 8 - (*lag_frac < 0),
313 AMRWB_P_DELAY_MIN, AMRWB_P_DELAY_MAX - 15);
315 *lag_int = (pitch_index + 1) >> 1;
316 *lag_frac = (pitch_index - (*lag_int << 1)) * 2;
317 *lag_int += *base_lag_int;
322 * Find the pitch vector by interpolating the past excitation at the
323 * pitch delay, which is obtained in this function.
325 * @param[in,out] ctx The context
326 * @param[in] amr_subframe Current subframe data
327 * @param[in] subframe Current subframe index (0 to 3)
329 static void decode_pitch_vector(AMRWBContext *ctx,
330 const AMRWBSubFrame *amr_subframe,
333 int pitch_lag_int, pitch_lag_frac;
335 float *exc = ctx->excitation;
336 enum Mode mode = ctx->fr_cur_mode;
338 if (mode <= MODE_8k85) {
339 decode_pitch_lag_low(&pitch_lag_int, &pitch_lag_frac, amr_subframe->adap,
340 &ctx->base_pitch_lag, subframe, mode);
342 decode_pitch_lag_high(&pitch_lag_int, &pitch_lag_frac, amr_subframe->adap,
343 &ctx->base_pitch_lag, subframe);
345 ctx->pitch_lag_int = pitch_lag_int;
346 pitch_lag_int += pitch_lag_frac > 0;
348 /* Calculate the pitch vector by interpolating the past excitation at the
349 pitch lag using a hamming windowed sinc function */
350 ctx->acelpf_ctx.acelp_interpolatef(exc,
351 exc + 1 - pitch_lag_int,
353 pitch_lag_frac + (pitch_lag_frac > 0 ? 0 : 4),
354 LP_ORDER, AMRWB_SFR_SIZE + 1);
356 /* Check which pitch signal path should be used
357 * 6k60 and 8k85 modes have the ltp flag set to 0 */
358 if (amr_subframe->ltp) {
359 memcpy(ctx->pitch_vector, exc, AMRWB_SFR_SIZE * sizeof(float));
361 for (i = 0; i < AMRWB_SFR_SIZE; i++)
362 ctx->pitch_vector[i] = 0.18 * exc[i - 1] + 0.64 * exc[i] +
364 memcpy(exc, ctx->pitch_vector, AMRWB_SFR_SIZE * sizeof(float));
368 /** Get x bits in the index interval [lsb,lsb+len-1] inclusive */
369 #define BIT_STR(x,lsb,len) av_mod_uintp2((x) >> (lsb), (len))
371 /** Get the bit at specified position */
372 #define BIT_POS(x, p) (((x) >> (p)) & 1)
375 * The next six functions decode_[i]p_track decode exactly i pulses
376 * positions and amplitudes (-1 or 1) in a subframe track using
377 * an encoded pulse indexing (TS 26.190 section 5.8.2).
379 * The results are given in out[], in which a negative number means
380 * amplitude -1 and vice versa (i.e., ampl(x) = x / abs(x) ).
382 * @param[out] out Output buffer (writes i elements)
383 * @param[in] code Pulse index (no. of bits varies, see below)
384 * @param[in] m (log2) Number of potential positions
385 * @param[in] off Offset for decoded positions
387 static inline void decode_1p_track(int *out, int code, int m, int off)
389 int pos = BIT_STR(code, 0, m) + off; ///code: m+1 bits
391 out[0] = BIT_POS(code, m) ? -pos : pos;
394 static inline void decode_2p_track(int *out, int code, int m, int off) ///code: 2m+1 bits
396 int pos0 = BIT_STR(code, m, m) + off;
397 int pos1 = BIT_STR(code, 0, m) + off;
399 out[0] = BIT_POS(code, 2*m) ? -pos0 : pos0;
400 out[1] = BIT_POS(code, 2*m) ? -pos1 : pos1;
401 out[1] = pos0 > pos1 ? -out[1] : out[1];
404 static void decode_3p_track(int *out, int code, int m, int off) ///code: 3m+1 bits
406 int half_2p = BIT_POS(code, 2*m - 1) << (m - 1);
408 decode_2p_track(out, BIT_STR(code, 0, 2*m - 1),
409 m - 1, off + half_2p);
410 decode_1p_track(out + 2, BIT_STR(code, 2*m, m + 1), m, off);
413 static void decode_4p_track(int *out, int code, int m, int off) ///code: 4m bits
415 int half_4p, subhalf_2p;
416 int b_offset = 1 << (m - 1);
418 switch (BIT_STR(code, 4*m - 2, 2)) { /* case ID (2 bits) */
419 case 0: /* 0 pulses in A, 4 pulses in B or vice versa */
420 half_4p = BIT_POS(code, 4*m - 3) << (m - 1); // which has 4 pulses
421 subhalf_2p = BIT_POS(code, 2*m - 3) << (m - 2);
423 decode_2p_track(out, BIT_STR(code, 0, 2*m - 3),
424 m - 2, off + half_4p + subhalf_2p);
425 decode_2p_track(out + 2, BIT_STR(code, 2*m - 2, 2*m - 1),
426 m - 1, off + half_4p);
428 case 1: /* 1 pulse in A, 3 pulses in B */
429 decode_1p_track(out, BIT_STR(code, 3*m - 2, m),
431 decode_3p_track(out + 1, BIT_STR(code, 0, 3*m - 2),
432 m - 1, off + b_offset);
434 case 2: /* 2 pulses in each half */
435 decode_2p_track(out, BIT_STR(code, 2*m - 1, 2*m - 1),
437 decode_2p_track(out + 2, BIT_STR(code, 0, 2*m - 1),
438 m - 1, off + b_offset);
440 case 3: /* 3 pulses in A, 1 pulse in B */
441 decode_3p_track(out, BIT_STR(code, m, 3*m - 2),
443 decode_1p_track(out + 3, BIT_STR(code, 0, m),
444 m - 1, off + b_offset);
449 static void decode_5p_track(int *out, int code, int m, int off) ///code: 5m bits
451 int half_3p = BIT_POS(code, 5*m - 1) << (m - 1);
453 decode_3p_track(out, BIT_STR(code, 2*m + 1, 3*m - 2),
454 m - 1, off + half_3p);
456 decode_2p_track(out + 3, BIT_STR(code, 0, 2*m + 1), m, off);
459 static void decode_6p_track(int *out, int code, int m, int off) ///code: 6m-2 bits
461 int b_offset = 1 << (m - 1);
462 /* which half has more pulses in cases 0 to 2 */
463 int half_more = BIT_POS(code, 6*m - 5) << (m - 1);
464 int half_other = b_offset - half_more;
466 switch (BIT_STR(code, 6*m - 4, 2)) { /* case ID (2 bits) */
467 case 0: /* 0 pulses in A, 6 pulses in B or vice versa */
468 decode_1p_track(out, BIT_STR(code, 0, m),
469 m - 1, off + half_more);
470 decode_5p_track(out + 1, BIT_STR(code, m, 5*m - 5),
471 m - 1, off + half_more);
473 case 1: /* 1 pulse in A, 5 pulses in B or vice versa */
474 decode_1p_track(out, BIT_STR(code, 0, m),
475 m - 1, off + half_other);
476 decode_5p_track(out + 1, BIT_STR(code, m, 5*m - 5),
477 m - 1, off + half_more);
479 case 2: /* 2 pulses in A, 4 pulses in B or vice versa */
480 decode_2p_track(out, BIT_STR(code, 0, 2*m - 1),
481 m - 1, off + half_other);
482 decode_4p_track(out + 2, BIT_STR(code, 2*m - 1, 4*m - 4),
483 m - 1, off + half_more);
485 case 3: /* 3 pulses in A, 3 pulses in B */
486 decode_3p_track(out, BIT_STR(code, 3*m - 2, 3*m - 2),
488 decode_3p_track(out + 3, BIT_STR(code, 0, 3*m - 2),
489 m - 1, off + b_offset);
495 * Decode the algebraic codebook index to pulse positions and signs,
496 * then construct the algebraic codebook vector.
498 * @param[out] fixed_vector Buffer for the fixed codebook excitation
499 * @param[in] pulse_hi MSBs part of the pulse index array (higher modes only)
500 * @param[in] pulse_lo LSBs part of the pulse index array
501 * @param[in] mode Mode of the current frame
503 static void decode_fixed_vector(float *fixed_vector, const uint16_t *pulse_hi,
504 const uint16_t *pulse_lo, const enum Mode mode)
506 /* sig_pos stores for each track the decoded pulse position indexes
507 * (1-based) multiplied by its corresponding amplitude (+1 or -1) */
509 int spacing = (mode == MODE_6k60) ? 2 : 4;
514 for (i = 0; i < 2; i++)
515 decode_1p_track(sig_pos[i], pulse_lo[i], 5, 1);
518 for (i = 0; i < 4; i++)
519 decode_1p_track(sig_pos[i], pulse_lo[i], 4, 1);
522 for (i = 0; i < 4; i++)
523 decode_2p_track(sig_pos[i], pulse_lo[i], 4, 1);
526 for (i = 0; i < 2; i++)
527 decode_3p_track(sig_pos[i], pulse_lo[i], 4, 1);
528 for (i = 2; i < 4; i++)
529 decode_2p_track(sig_pos[i], pulse_lo[i], 4, 1);
532 for (i = 0; i < 4; i++)
533 decode_3p_track(sig_pos[i], pulse_lo[i], 4, 1);
536 for (i = 0; i < 4; i++)
537 decode_4p_track(sig_pos[i], (int) pulse_lo[i] +
538 ((int) pulse_hi[i] << 14), 4, 1);
541 for (i = 0; i < 2; i++)
542 decode_5p_track(sig_pos[i], (int) pulse_lo[i] +
543 ((int) pulse_hi[i] << 10), 4, 1);
544 for (i = 2; i < 4; i++)
545 decode_4p_track(sig_pos[i], (int) pulse_lo[i] +
546 ((int) pulse_hi[i] << 14), 4, 1);
550 for (i = 0; i < 4; i++)
551 decode_6p_track(sig_pos[i], (int) pulse_lo[i] +
552 ((int) pulse_hi[i] << 11), 4, 1);
556 memset(fixed_vector, 0, sizeof(float) * AMRWB_SFR_SIZE);
558 for (i = 0; i < 4; i++)
559 for (j = 0; j < pulses_nb_per_mode_tr[mode][i]; j++) {
560 int pos = (FFABS(sig_pos[i][j]) - 1) * spacing + i;
562 fixed_vector[pos] += sig_pos[i][j] < 0 ? -1.0 : 1.0;
567 * Decode pitch gain and fixed gain correction factor.
569 * @param[in] vq_gain Vector-quantized index for gains
570 * @param[in] mode Mode of the current frame
571 * @param[out] fixed_gain_factor Decoded fixed gain correction factor
572 * @param[out] pitch_gain Decoded pitch gain
574 static void decode_gains(const uint8_t vq_gain, const enum Mode mode,
575 float *fixed_gain_factor, float *pitch_gain)
577 const int16_t *gains = (mode <= MODE_8k85 ? qua_gain_6b[vq_gain] :
578 qua_gain_7b[vq_gain]);
580 *pitch_gain = gains[0] * (1.0f / (1 << 14));
581 *fixed_gain_factor = gains[1] * (1.0f / (1 << 11));
585 * Apply pitch sharpening filters to the fixed codebook vector.
587 * @param[in] ctx The context
588 * @param[in,out] fixed_vector Fixed codebook excitation
590 // XXX: Spec states this procedure should be applied when the pitch
591 // lag is less than 64, but this checking seems absent in reference and AMR-NB
592 static void pitch_sharpening(AMRWBContext *ctx, float *fixed_vector)
597 for (i = AMRWB_SFR_SIZE - 1; i != 0; i--)
598 fixed_vector[i] -= fixed_vector[i - 1] * ctx->tilt_coef;
600 /* Periodicity enhancement part */
601 for (i = ctx->pitch_lag_int; i < AMRWB_SFR_SIZE; i++)
602 fixed_vector[i] += fixed_vector[i - ctx->pitch_lag_int] * 0.85;
606 * Calculate the voicing factor (-1.0 = unvoiced to 1.0 = voiced).
608 * @param[in] p_vector, f_vector Pitch and fixed excitation vectors
609 * @param[in] p_gain, f_gain Pitch and fixed gains
610 * @param[in] ctx The context
612 // XXX: There is something wrong with the precision here! The magnitudes
613 // of the energies are not correct. Please check the reference code carefully
614 static float voice_factor(float *p_vector, float p_gain,
615 float *f_vector, float f_gain,
618 double p_ener = (double) ctx->dot_productf(p_vector, p_vector,
621 double f_ener = (double) ctx->dot_productf(f_vector, f_vector,
625 return (p_ener - f_ener) / (p_ener + f_ener + 0.01);
629 * Reduce fixed vector sparseness by smoothing with one of three IR filters,
630 * also known as "adaptive phase dispersion".
632 * @param[in] ctx The context
633 * @param[in,out] fixed_vector Unfiltered fixed vector
634 * @param[out] buf Space for modified vector if necessary
636 * @return The potentially overwritten filtered fixed vector address
638 static float *anti_sparseness(AMRWBContext *ctx,
639 float *fixed_vector, float *buf)
643 if (ctx->fr_cur_mode > MODE_8k85) // no filtering in higher modes
646 if (ctx->pitch_gain[0] < 0.6) {
647 ir_filter_nr = 0; // strong filtering
648 } else if (ctx->pitch_gain[0] < 0.9) {
649 ir_filter_nr = 1; // medium filtering
651 ir_filter_nr = 2; // no filtering
654 if (ctx->fixed_gain[0] > 3.0 * ctx->fixed_gain[1]) {
655 if (ir_filter_nr < 2)
660 for (i = 0; i < 6; i++)
661 if (ctx->pitch_gain[i] < 0.6)
667 if (ir_filter_nr > ctx->prev_ir_filter_nr + 1)
671 /* update ir filter strength history */
672 ctx->prev_ir_filter_nr = ir_filter_nr;
674 ir_filter_nr += (ctx->fr_cur_mode == MODE_8k85);
676 if (ir_filter_nr < 2) {
678 const float *coef = ir_filters_lookup[ir_filter_nr];
680 /* Circular convolution code in the reference
681 * decoder was modified to avoid using one
682 * extra array. The filtered vector is given by:
684 * c2(n) = sum(i,0,len-1){ c(i) * coef( (n - i + len) % len ) }
687 memset(buf, 0, sizeof(float) * AMRWB_SFR_SIZE);
688 for (i = 0; i < AMRWB_SFR_SIZE; i++)
690 ff_celp_circ_addf(buf, buf, coef, i, fixed_vector[i],
699 * Calculate a stability factor {teta} based on distance between
700 * current and past isf. A value of 1 shows maximum signal stability.
702 static float stability_factor(const float *isf, const float *isf_past)
707 for (i = 0; i < LP_ORDER - 1; i++)
708 acc += (isf[i] - isf_past[i]) * (isf[i] - isf_past[i]);
710 // XXX: This part is not so clear from the reference code
711 // the result is more accurate changing the "/ 256" to "* 512"
712 return FFMAX(0.0, 1.25 - acc * 0.8 * 512);
716 * Apply a non-linear fixed gain smoothing in order to reduce
717 * fluctuation in the energy of excitation.
719 * @param[in] fixed_gain Unsmoothed fixed gain
720 * @param[in,out] prev_tr_gain Previous threshold gain (updated)
721 * @param[in] voice_fac Frame voicing factor
722 * @param[in] stab_fac Frame stability factor
724 * @return The smoothed gain
726 static float noise_enhancer(float fixed_gain, float *prev_tr_gain,
727 float voice_fac, float stab_fac)
729 float sm_fac = 0.5 * (1 - voice_fac) * stab_fac;
732 // XXX: the following fixed-point constants used to in(de)crement
733 // gain by 1.5dB were taken from the reference code, maybe it could
735 if (fixed_gain < *prev_tr_gain) {
736 g0 = FFMIN(*prev_tr_gain, fixed_gain + fixed_gain *
737 (6226 * (1.0f / (1 << 15)))); // +1.5 dB
739 g0 = FFMAX(*prev_tr_gain, fixed_gain *
740 (27536 * (1.0f / (1 << 15)))); // -1.5 dB
742 *prev_tr_gain = g0; // update next frame threshold
744 return sm_fac * g0 + (1 - sm_fac) * fixed_gain;
748 * Filter the fixed_vector to emphasize the higher frequencies.
750 * @param[in,out] fixed_vector Fixed codebook vector
751 * @param[in] voice_fac Frame voicing factor
753 static void pitch_enhancer(float *fixed_vector, float voice_fac)
756 float cpe = 0.125 * (1 + voice_fac);
757 float last = fixed_vector[0]; // holds c(i - 1)
759 fixed_vector[0] -= cpe * fixed_vector[1];
761 for (i = 1; i < AMRWB_SFR_SIZE - 1; i++) {
762 float cur = fixed_vector[i];
764 fixed_vector[i] -= cpe * (last + fixed_vector[i + 1]);
768 fixed_vector[AMRWB_SFR_SIZE - 1] -= cpe * last;
772 * Conduct 16th order linear predictive coding synthesis from excitation.
774 * @param[in] ctx Pointer to the AMRWBContext
775 * @param[in] lpc Pointer to the LPC coefficients
776 * @param[out] excitation Buffer for synthesis final excitation
777 * @param[in] fixed_gain Fixed codebook gain for synthesis
778 * @param[in] fixed_vector Algebraic codebook vector
779 * @param[in,out] samples Pointer to the output samples and memory
781 static void synthesis(AMRWBContext *ctx, float *lpc, float *excitation,
782 float fixed_gain, const float *fixed_vector,
785 ctx->acelpv_ctx.weighted_vector_sumf(excitation, ctx->pitch_vector, fixed_vector,
786 ctx->pitch_gain[0], fixed_gain, AMRWB_SFR_SIZE);
788 /* emphasize pitch vector contribution in low bitrate modes */
789 if (ctx->pitch_gain[0] > 0.5 && ctx->fr_cur_mode <= MODE_8k85) {
791 float energy = ctx->celpm_ctx.dot_productf(excitation, excitation,
794 // XXX: Weird part in both ref code and spec. A unknown parameter
795 // {beta} seems to be identical to the current pitch gain
796 float pitch_factor = 0.25 * ctx->pitch_gain[0] * ctx->pitch_gain[0];
798 for (i = 0; i < AMRWB_SFR_SIZE; i++)
799 excitation[i] += pitch_factor * ctx->pitch_vector[i];
801 ff_scale_vector_to_given_sum_of_squares(excitation, excitation,
802 energy, AMRWB_SFR_SIZE);
805 ctx->celpf_ctx.celp_lp_synthesis_filterf(samples, lpc, excitation,
806 AMRWB_SFR_SIZE, LP_ORDER);
810 * Apply to synthesis a de-emphasis filter of the form:
811 * H(z) = 1 / (1 - m * z^-1)
813 * @param[out] out Output buffer
814 * @param[in] in Input samples array with in[-1]
815 * @param[in] m Filter coefficient
816 * @param[in,out] mem State from last filtering
818 static void de_emphasis(float *out, float *in, float m, float mem[1])
822 out[0] = in[0] + m * mem[0];
824 for (i = 1; i < AMRWB_SFR_SIZE; i++)
825 out[i] = in[i] + out[i - 1] * m;
827 mem[0] = out[AMRWB_SFR_SIZE - 1];
831 * Upsample a signal by 5/4 ratio (from 12.8kHz to 16kHz) using
832 * a FIR interpolation filter. Uses past data from before *in address.
834 * @param[out] out Buffer for interpolated signal
835 * @param[in] in Current signal data (length 0.8*o_size)
836 * @param[in] o_size Output signal length
837 * @param[in] ctx The context
839 static void upsample_5_4(float *out, const float *in, int o_size, CELPMContext *ctx)
841 const float *in0 = in - UPS_FIR_SIZE + 1;
843 int int_part = 0, frac_part;
846 for (j = 0; j < o_size / 5; j++) {
847 out[i] = in[int_part];
851 for (k = 1; k < 5; k++) {
852 out[i] = ctx->dot_productf(in0 + int_part,
853 upsample_fir[4 - frac_part],
863 * Calculate the high-band gain based on encoded index (23k85 mode) or
864 * on the low-band speech signal and the Voice Activity Detection flag.
866 * @param[in] ctx The context
867 * @param[in] synth LB speech synthesis at 12.8k
868 * @param[in] hb_idx Gain index for mode 23k85 only
869 * @param[in] vad VAD flag for the frame
871 static float find_hb_gain(AMRWBContext *ctx, const float *synth,
872 uint16_t hb_idx, uint8_t vad)
878 if (ctx->fr_cur_mode == MODE_23k85)
879 return qua_hb_gain[hb_idx] * (1.0f / (1 << 14));
881 tmp = ctx->celpm_ctx.dot_productf(synth, synth + 1, AMRWB_SFR_SIZE - 1);
884 tilt = tmp / ctx->celpm_ctx.dot_productf(synth, synth, AMRWB_SFR_SIZE);
888 /* return gain bounded by [0.1, 1.0] */
889 return av_clipf((1.0 - tilt) * (1.25 - 0.25 * wsp), 0.1, 1.0);
893 * Generate the high-band excitation with the same energy from the lower
894 * one and scaled by the given gain.
896 * @param[in] ctx The context
897 * @param[out] hb_exc Buffer for the excitation
898 * @param[in] synth_exc Low-band excitation used for synthesis
899 * @param[in] hb_gain Wanted excitation gain
901 static void scaled_hb_excitation(AMRWBContext *ctx, float *hb_exc,
902 const float *synth_exc, float hb_gain)
905 float energy = ctx->celpm_ctx.dot_productf(synth_exc, synth_exc,
908 /* Generate a white-noise excitation */
909 for (i = 0; i < AMRWB_SFR_SIZE_16k; i++)
910 hb_exc[i] = 32768.0 - (uint16_t) av_lfg_get(&ctx->prng);
912 ff_scale_vector_to_given_sum_of_squares(hb_exc, hb_exc,
913 energy * hb_gain * hb_gain,
918 * Calculate the auto-correlation for the ISF difference vector.
920 static float auto_correlation(float *diff_isf, float mean, int lag)
925 for (i = 7; i < LP_ORDER - 2; i++) {
926 float prod = (diff_isf[i] - mean) * (diff_isf[i - lag] - mean);
933 * Extrapolate a ISF vector to the 16kHz range (20th order LP)
934 * used at mode 6k60 LP filter for the high frequency band.
936 * @param[out] isf Buffer for extrapolated isf; contains LP_ORDER
939 static void extrapolate_isf(float isf[LP_ORDER_16k])
941 float diff_isf[LP_ORDER - 2], diff_mean;
944 int i, j, i_max_corr;
946 isf[LP_ORDER_16k - 1] = isf[LP_ORDER - 1];
948 /* Calculate the difference vector */
949 for (i = 0; i < LP_ORDER - 2; i++)
950 diff_isf[i] = isf[i + 1] - isf[i];
953 for (i = 2; i < LP_ORDER - 2; i++)
954 diff_mean += diff_isf[i] * (1.0f / (LP_ORDER - 4));
956 /* Find which is the maximum autocorrelation */
958 for (i = 0; i < 3; i++) {
959 corr_lag[i] = auto_correlation(diff_isf, diff_mean, i + 2);
961 if (corr_lag[i] > corr_lag[i_max_corr])
966 for (i = LP_ORDER - 1; i < LP_ORDER_16k - 1; i++)
967 isf[i] = isf[i - 1] + isf[i - 1 - i_max_corr]
968 - isf[i - 2 - i_max_corr];
970 /* Calculate an estimate for ISF(18) and scale ISF based on the error */
971 est = 7965 + (isf[2] - isf[3] - isf[4]) / 6.0;
972 scale = 0.5 * (FFMIN(est, 7600) - isf[LP_ORDER - 2]) /
973 (isf[LP_ORDER_16k - 2] - isf[LP_ORDER - 2]);
975 for (i = LP_ORDER - 1, j = 0; i < LP_ORDER_16k - 1; i++, j++)
976 diff_isf[j] = scale * (isf[i] - isf[i - 1]);
978 /* Stability insurance */
979 for (i = 1; i < LP_ORDER_16k - LP_ORDER; i++)
980 if (diff_isf[i] + diff_isf[i - 1] < 5.0) {
981 if (diff_isf[i] > diff_isf[i - 1]) {
982 diff_isf[i - 1] = 5.0 - diff_isf[i];
984 diff_isf[i] = 5.0 - diff_isf[i - 1];
987 for (i = LP_ORDER - 1, j = 0; i < LP_ORDER_16k - 1; i++, j++)
988 isf[i] = isf[i - 1] + diff_isf[j] * (1.0f / (1 << 15));
990 /* Scale the ISF vector for 16000 Hz */
991 for (i = 0; i < LP_ORDER_16k - 1; i++)
996 * Spectral expand the LP coefficients using the equation:
997 * y[i] = x[i] * (gamma ** i)
999 * @param[out] out Output buffer (may use input array)
1000 * @param[in] lpc LP coefficients array
1001 * @param[in] gamma Weighting factor
1002 * @param[in] size LP array size
1004 static void lpc_weighting(float *out, const float *lpc, float gamma, int size)
1009 for (i = 0; i < size; i++) {
1010 out[i] = lpc[i] * fac;
1016 * Conduct 20th order linear predictive coding synthesis for the high
1017 * frequency band excitation at 16kHz.
1019 * @param[in] ctx The context
1020 * @param[in] subframe Current subframe index (0 to 3)
1021 * @param[in,out] samples Pointer to the output speech samples
1022 * @param[in] exc Generated white-noise scaled excitation
1023 * @param[in] isf Current frame isf vector
1024 * @param[in] isf_past Past frame final isf vector
1026 static void hb_synthesis(AMRWBContext *ctx, int subframe, float *samples,
1027 const float *exc, const float *isf, const float *isf_past)
1029 float hb_lpc[LP_ORDER_16k];
1030 enum Mode mode = ctx->fr_cur_mode;
1032 if (mode == MODE_6k60) {
1033 float e_isf[LP_ORDER_16k]; // ISF vector for extrapolation
1034 double e_isp[LP_ORDER_16k];
1036 ctx->acelpv_ctx.weighted_vector_sumf(e_isf, isf_past, isf, isfp_inter[subframe],
1037 1.0 - isfp_inter[subframe], LP_ORDER);
1039 extrapolate_isf(e_isf);
1041 e_isf[LP_ORDER_16k - 1] *= 2.0;
1042 ff_acelp_lsf2lspd(e_isp, e_isf, LP_ORDER_16k);
1043 ff_amrwb_lsp2lpc(e_isp, hb_lpc, LP_ORDER_16k);
1045 lpc_weighting(hb_lpc, hb_lpc, 0.9, LP_ORDER_16k);
1047 lpc_weighting(hb_lpc, ctx->lp_coef[subframe], 0.6, LP_ORDER);
1050 ctx->celpf_ctx.celp_lp_synthesis_filterf(samples, hb_lpc, exc, AMRWB_SFR_SIZE_16k,
1051 (mode == MODE_6k60) ? LP_ORDER_16k : LP_ORDER);
1055 * Apply a 15th order filter to high-band samples.
1056 * The filter characteristic depends on the given coefficients.
1058 * @param[out] out Buffer for filtered output
1059 * @param[in] fir_coef Filter coefficients
1060 * @param[in,out] mem State from last filtering (updated)
1061 * @param[in] in Input speech data (high-band)
1063 * @remark It is safe to pass the same array in in and out parameters
1066 #ifndef hb_fir_filter
1067 static void hb_fir_filter(float *out, const float fir_coef[HB_FIR_SIZE + 1],
1068 float mem[HB_FIR_SIZE], const float *in)
1071 float data[AMRWB_SFR_SIZE_16k + HB_FIR_SIZE]; // past and current samples
1073 memcpy(data, mem, HB_FIR_SIZE * sizeof(float));
1074 memcpy(data + HB_FIR_SIZE, in, AMRWB_SFR_SIZE_16k * sizeof(float));
1076 for (i = 0; i < AMRWB_SFR_SIZE_16k; i++) {
1078 for (j = 0; j <= HB_FIR_SIZE; j++)
1079 out[i] += data[i + j] * fir_coef[j];
1082 memcpy(mem, data + AMRWB_SFR_SIZE_16k, HB_FIR_SIZE * sizeof(float));
1084 #endif /* hb_fir_filter */
1087 * Update context state before the next subframe.
1089 static void update_sub_state(AMRWBContext *ctx)
1091 memmove(&ctx->excitation_buf[0], &ctx->excitation_buf[AMRWB_SFR_SIZE],
1092 (AMRWB_P_DELAY_MAX + LP_ORDER + 1) * sizeof(float));
1094 memmove(&ctx->pitch_gain[1], &ctx->pitch_gain[0], 5 * sizeof(float));
1095 memmove(&ctx->fixed_gain[1], &ctx->fixed_gain[0], 1 * sizeof(float));
1097 memmove(&ctx->samples_az[0], &ctx->samples_az[AMRWB_SFR_SIZE],
1098 LP_ORDER * sizeof(float));
1099 memmove(&ctx->samples_up[0], &ctx->samples_up[AMRWB_SFR_SIZE],
1100 UPS_MEM_SIZE * sizeof(float));
1101 memmove(&ctx->samples_hb[0], &ctx->samples_hb[AMRWB_SFR_SIZE_16k],
1102 LP_ORDER_16k * sizeof(float));
1105 static int amrwb_decode_frame(AVCodecContext *avctx, AVFrame *frame,
1106 int *got_frame_ptr, AVPacket *avpkt)
1108 AMRWBChannelsContext *s = avctx->priv_data;
1109 const uint8_t *buf = avpkt->data;
1110 int buf_size = avpkt->size;
1113 /* get output buffer */
1114 frame->nb_samples = 4 * AMRWB_SFR_SIZE_16k;
1115 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
1118 for (int ch = 0; ch < avctx->ch_layout.nb_channels; ch++) {
1119 AMRWBContext *ctx = &s->ch[ch];
1120 AMRWBFrame *cf = &ctx->frame;
1121 int expected_fr_size, header_size;
1122 float spare_vector[AMRWB_SFR_SIZE]; // extra stack space to hold result from anti-sparseness processing
1123 float fixed_gain_factor; // fixed gain correction factor (gamma)
1124 float *synth_fixed_vector; // pointer to the fixed vector that synthesis should use
1125 float synth_fixed_gain; // the fixed gain that synthesis should use
1126 float voice_fac, stab_fac; // parameters used for gain smoothing
1127 float synth_exc[AMRWB_SFR_SIZE]; // post-processed excitation for synthesis
1128 float hb_exc[AMRWB_SFR_SIZE_16k]; // excitation for the high frequency band
1129 float hb_samples[AMRWB_SFR_SIZE_16k]; // filtered high-band samples from synthesis
1131 float *buf_out = (float *)frame->extended_data[ch];
1133 header_size = decode_mime_header(ctx, buf);
1134 expected_fr_size = ((cf_sizes_wb[ctx->fr_cur_mode] + 7) >> 3) + 1;
1136 if (!ctx->fr_quality)
1137 av_log(avctx, AV_LOG_ERROR, "Encountered a bad or corrupted frame\n");
1139 if (ctx->fr_cur_mode == NO_DATA || !ctx->fr_quality) {
1140 /* The specification suggests a "random signal" and
1141 "a muting technique" to "gradually decrease the output level". */
1142 av_samples_set_silence(&frame->extended_data[ch], 0, frame->nb_samples, 1, AV_SAMPLE_FMT_FLT);
1143 buf += expected_fr_size;
1144 buf_size -= expected_fr_size;
1147 if (ctx->fr_cur_mode > MODE_SID) {
1148 av_log(avctx, AV_LOG_ERROR,
1149 "Invalid mode %d\n", ctx->fr_cur_mode);
1150 return AVERROR_INVALIDDATA;
1153 if (buf_size < expected_fr_size) {
1154 av_log(avctx, AV_LOG_ERROR,
1155 "Frame too small (%d bytes). Truncated file?\n", buf_size);
1157 return AVERROR_INVALIDDATA;
1160 if (ctx->fr_cur_mode == MODE_SID) { /* Comfort noise frame */
1161 avpriv_request_sample(avctx, "SID mode");
1162 return AVERROR_PATCHWELCOME;
1165 ff_amr_bit_reorder((uint16_t *) &ctx->frame, sizeof(AMRWBFrame),
1166 buf + header_size, amr_bit_orderings_by_mode[ctx->fr_cur_mode]);
1168 /* Decode the quantized ISF vector */
1169 if (ctx->fr_cur_mode == MODE_6k60) {
1170 decode_isf_indices_36b(cf->isp_id, ctx->isf_cur);
1172 decode_isf_indices_46b(cf->isp_id, ctx->isf_cur);
1175 isf_add_mean_and_past(ctx->isf_cur, ctx->isf_q_past);
1176 ff_set_min_dist_lsf(ctx->isf_cur, MIN_ISF_SPACING, LP_ORDER - 1);
1178 stab_fac = stability_factor(ctx->isf_cur, ctx->isf_past_final);
1180 ctx->isf_cur[LP_ORDER - 1] *= 2.0;
1181 ff_acelp_lsf2lspd(ctx->isp[3], ctx->isf_cur, LP_ORDER);
1183 /* Generate a ISP vector for each subframe */
1184 if (ctx->first_frame) {
1185 ctx->first_frame = 0;
1186 memcpy(ctx->isp_sub4_past, ctx->isp[3], LP_ORDER * sizeof(double));
1188 interpolate_isp(ctx->isp, ctx->isp_sub4_past);
1190 for (sub = 0; sub < 4; sub++)
1191 ff_amrwb_lsp2lpc(ctx->isp[sub], ctx->lp_coef[sub], LP_ORDER);
1193 for (sub = 0; sub < 4; sub++) {
1194 const AMRWBSubFrame *cur_subframe = &cf->subframe[sub];
1195 float *sub_buf = buf_out + sub * AMRWB_SFR_SIZE_16k;
1197 /* Decode adaptive codebook (pitch vector) */
1198 decode_pitch_vector(ctx, cur_subframe, sub);
1199 /* Decode innovative codebook (fixed vector) */
1200 decode_fixed_vector(ctx->fixed_vector, cur_subframe->pul_ih,
1201 cur_subframe->pul_il, ctx->fr_cur_mode);
1203 pitch_sharpening(ctx, ctx->fixed_vector);
1205 decode_gains(cur_subframe->vq_gain, ctx->fr_cur_mode,
1206 &fixed_gain_factor, &ctx->pitch_gain[0]);
1208 ctx->fixed_gain[0] =
1209 ff_amr_set_fixed_gain(fixed_gain_factor,
1210 ctx->celpm_ctx.dot_productf(ctx->fixed_vector,
1214 ctx->prediction_error,
1215 ENERGY_MEAN, energy_pred_fac);
1217 /* Calculate voice factor and store tilt for next subframe */
1218 voice_fac = voice_factor(ctx->pitch_vector, ctx->pitch_gain[0],
1219 ctx->fixed_vector, ctx->fixed_gain[0],
1221 ctx->tilt_coef = voice_fac * 0.25 + 0.25;
1223 /* Construct current excitation */
1224 for (i = 0; i < AMRWB_SFR_SIZE; i++) {
1225 ctx->excitation[i] *= ctx->pitch_gain[0];
1226 ctx->excitation[i] += ctx->fixed_gain[0] * ctx->fixed_vector[i];
1227 ctx->excitation[i] = truncf(ctx->excitation[i]);
1230 /* Post-processing of excitation elements */
1231 synth_fixed_gain = noise_enhancer(ctx->fixed_gain[0], &ctx->prev_tr_gain,
1232 voice_fac, stab_fac);
1234 synth_fixed_vector = anti_sparseness(ctx, ctx->fixed_vector,
1237 pitch_enhancer(synth_fixed_vector, voice_fac);
1239 synthesis(ctx, ctx->lp_coef[sub], synth_exc, synth_fixed_gain,
1240 synth_fixed_vector, &ctx->samples_az[LP_ORDER]);
1242 /* Synthesis speech post-processing */
1243 de_emphasis(&ctx->samples_up[UPS_MEM_SIZE],
1244 &ctx->samples_az[LP_ORDER], PREEMPH_FAC, ctx->demph_mem);
1246 ctx->acelpf_ctx.acelp_apply_order_2_transfer_function(&ctx->samples_up[UPS_MEM_SIZE],
1247 &ctx->samples_up[UPS_MEM_SIZE], hpf_zeros, hpf_31_poles,
1248 hpf_31_gain, ctx->hpf_31_mem, AMRWB_SFR_SIZE);
1250 upsample_5_4(sub_buf, &ctx->samples_up[UPS_FIR_SIZE],
1251 AMRWB_SFR_SIZE_16k, &ctx->celpm_ctx);
1253 /* High frequency band (6.4 - 7.0 kHz) generation part */
1254 ctx->acelpf_ctx.acelp_apply_order_2_transfer_function(hb_samples,
1255 &ctx->samples_up[UPS_MEM_SIZE], hpf_zeros, hpf_400_poles,
1256 hpf_400_gain, ctx->hpf_400_mem, AMRWB_SFR_SIZE);
1258 hb_gain = find_hb_gain(ctx, hb_samples,
1259 cur_subframe->hb_gain, cf->vad);
1261 scaled_hb_excitation(ctx, hb_exc, synth_exc, hb_gain);
1263 hb_synthesis(ctx, sub, &ctx->samples_hb[LP_ORDER_16k],
1264 hb_exc, ctx->isf_cur, ctx->isf_past_final);
1266 /* High-band post-processing filters */
1267 hb_fir_filter(hb_samples, bpf_6_7_coef, ctx->bpf_6_7_mem,
1268 &ctx->samples_hb[LP_ORDER_16k]);
1270 if (ctx->fr_cur_mode == MODE_23k85)
1271 hb_fir_filter(hb_samples, lpf_7_coef, ctx->lpf_7_mem,
1274 /* Add the low and high frequency bands */
1275 for (i = 0; i < AMRWB_SFR_SIZE_16k; i++)
1276 sub_buf[i] = (sub_buf[i] + hb_samples[i]) * (1.0f / (1 << 15));
1278 /* Update buffers and history */
1279 update_sub_state(ctx);
1282 /* update state for next frame */
1283 memcpy(ctx->isp_sub4_past, ctx->isp[3], LP_ORDER * sizeof(ctx->isp[3][0]));
1284 memcpy(ctx->isf_past_final, ctx->isf_cur, LP_ORDER * sizeof(float));
1286 buf += expected_fr_size;
1287 buf_size -= expected_fr_size;
1295 const FFCodec ff_amrwb_decoder = {
1297 .p.long_name = NULL_IF_CONFIG_SMALL("AMR-WB (Adaptive Multi-Rate WideBand)"),
1298 .p.type = AVMEDIA_TYPE_AUDIO,
1299 .p.id = AV_CODEC_ID_AMR_WB,
1300 .priv_data_size = sizeof(AMRWBChannelsContext),
1301 .init = amrwb_decode_init,
1302 FF_CODEC_DECODE_CB(amrwb_decode_frame),
1303 .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF,
1304 .p.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT,
1305 AV_SAMPLE_FMT_NONE },
1306 .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,