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"
41 #define AMR_USE_16BIT_TABLES
44 #include "amrwbdata.h"
45 #include "mips/amrwbdec_mips.h"
48 AMRWBFrame frame
; ///< AMRWB parameters decoded from bitstream
49 enum Mode fr_cur_mode
; ///< mode index of current frame
50 uint8_t fr_quality
; ///< frame quality index (FQI)
51 float isf_cur
[LP_ORDER
]; ///< working ISF vector from current frame
52 float isf_q_past
[LP_ORDER
]; ///< quantized ISF vector of the previous frame
53 float isf_past_final
[LP_ORDER
]; ///< final processed ISF vector of the previous frame
54 double isp
[4][LP_ORDER
]; ///< ISP vectors from current frame
55 double isp_sub4_past
[LP_ORDER
]; ///< ISP vector for the 4th subframe of the previous frame
57 float lp_coef
[4][LP_ORDER
]; ///< Linear Prediction Coefficients from ISP vector
59 uint8_t base_pitch_lag
; ///< integer part of pitch lag for the next relative subframe
60 uint8_t pitch_lag_int
; ///< integer part of pitch lag of the previous subframe
62 float excitation_buf
[AMRWB_P_DELAY_MAX
+ LP_ORDER
+ 2 + AMRWB_SFR_SIZE
]; ///< current excitation and all necessary excitation history
63 float *excitation
; ///< points to current excitation in excitation_buf[]
65 float pitch_vector
[AMRWB_SFR_SIZE
]; ///< adaptive codebook (pitch) vector for current subframe
66 float fixed_vector
[AMRWB_SFR_SIZE
]; ///< algebraic codebook (fixed) vector for current subframe
68 float prediction_error
[4]; ///< quantified prediction errors {20log10(^gamma_gc)} for previous four subframes
69 float pitch_gain
[6]; ///< quantified pitch gains for the current and previous five subframes
70 float fixed_gain
[2]; ///< quantified fixed gains for the current and previous subframes
72 float tilt_coef
; ///< {beta_1} related to the voicing of the previous subframe
74 float prev_sparse_fixed_gain
; ///< previous fixed gain; used by anti-sparseness to determine "onset"
75 uint8_t prev_ir_filter_nr
; ///< previous impulse response filter "impNr": 0 - strong, 1 - medium, 2 - none
76 float prev_tr_gain
; ///< previous initial gain used by noise enhancer for threshold
78 float samples_az
[LP_ORDER
+ AMRWB_SFR_SIZE
]; ///< low-band samples and memory from synthesis at 12.8kHz
79 float samples_up
[UPS_MEM_SIZE
+ AMRWB_SFR_SIZE
]; ///< low-band samples and memory processed for upsampling
80 float samples_hb
[LP_ORDER_16k
+ AMRWB_SFR_SIZE_16k
]; ///< high-band samples and memory from synthesis at 16kHz
82 float hpf_31_mem
[2], hpf_400_mem
[2]; ///< previous values in the high pass filters
83 float demph_mem
[1]; ///< previous value in the de-emphasis filter
84 float bpf_6_7_mem
[HB_FIR_SIZE
]; ///< previous values in the high-band band pass filter
85 float lpf_7_mem
[HB_FIR_SIZE
]; ///< previous values in the high-band low pass filter
87 AVLFG prng
; ///< random number generator for white noise excitation
88 uint8_t first_frame
; ///< flag active during decoding of the first frame
89 ACELPFContext acelpf_ctx
; ///< context for filters for ACELP-based codecs
90 ACELPVContext acelpv_ctx
; ///< context for vector operations for ACELP-based codecs
91 CELPFContext celpf_ctx
; ///< context for filters for CELP-based codecs
92 CELPMContext celpm_ctx
; ///< context for fixed point math operations
96 static av_cold
int amrwb_decode_init(AVCodecContext
*avctx
)
98 AMRWBContext
*ctx
= avctx
->priv_data
;
101 if (avctx
->channels
> 1) {
102 avpriv_report_missing_feature(avctx
, "multi-channel AMR");
103 return AVERROR_PATCHWELCOME
;
107 avctx
->channel_layout
= AV_CH_LAYOUT_MONO
;
108 if (!avctx
->sample_rate
)
109 avctx
->sample_rate
= 16000;
110 avctx
->sample_fmt
= AV_SAMPLE_FMT_FLT
;
112 av_lfg_init(&ctx
->prng
, 1);
114 ctx
->excitation
= &ctx
->excitation_buf
[AMRWB_P_DELAY_MAX
+ LP_ORDER
+ 1];
115 ctx
->first_frame
= 1;
117 for (i
= 0; i
< LP_ORDER
; i
++)
118 ctx
->isf_past_final
[i
] = isf_init
[i
] * (1.0f
/ (1 << 15));
120 for (i
= 0; i
< 4; i
++)
121 ctx
->prediction_error
[i
] = MIN_ENERGY
;
123 ff_acelp_filter_init(&ctx
->acelpf_ctx
);
124 ff_acelp_vectors_init(&ctx
->acelpv_ctx
);
125 ff_celp_filter_init(&ctx
->celpf_ctx
);
126 ff_celp_math_init(&ctx
->celpm_ctx
);
132 * Decode the frame header in the "MIME/storage" format. This format
133 * is simpler and does not carry the auxiliary frame information.
135 * @param[in] ctx The Context
136 * @param[in] buf Pointer to the input buffer
138 * @return The decoded header length in bytes
140 static int decode_mime_header(AMRWBContext
*ctx
, const uint8_t *buf
)
142 /* Decode frame header (1st octet) */
143 ctx
->fr_cur_mode
= buf
[0] >> 3 & 0x0F;
144 ctx
->fr_quality
= (buf
[0] & 0x4) == 0x4;
150 * Decode quantized ISF vectors using 36-bit indexes (6K60 mode only).
152 * @param[in] ind Array of 5 indexes
153 * @param[out] isf_q Buffer for isf_q[LP_ORDER]
156 static void decode_isf_indices_36b(uint16_t *ind
, float *isf_q
)
160 for (i
= 0; i
< 9; i
++)
161 isf_q
[i
] = dico1_isf
[ind
[0]][i
] * (1.0f
/ (1 << 15));
163 for (i
= 0; i
< 7; i
++)
164 isf_q
[i
+ 9] = dico2_isf
[ind
[1]][i
] * (1.0f
/ (1 << 15));
166 for (i
= 0; i
< 5; i
++)
167 isf_q
[i
] += dico21_isf_36b
[ind
[2]][i
] * (1.0f
/ (1 << 15));
169 for (i
= 0; i
< 4; i
++)
170 isf_q
[i
+ 5] += dico22_isf_36b
[ind
[3]][i
] * (1.0f
/ (1 << 15));
172 for (i
= 0; i
< 7; i
++)
173 isf_q
[i
+ 9] += dico23_isf_36b
[ind
[4]][i
] * (1.0f
/ (1 << 15));
177 * Decode quantized ISF vectors using 46-bit indexes (except 6K60 mode).
179 * @param[in] ind Array of 7 indexes
180 * @param[out] isf_q Buffer for isf_q[LP_ORDER]
183 static void decode_isf_indices_46b(uint16_t *ind
, float *isf_q
)
187 for (i
= 0; i
< 9; i
++)
188 isf_q
[i
] = dico1_isf
[ind
[0]][i
] * (1.0f
/ (1 << 15));
190 for (i
= 0; i
< 7; i
++)
191 isf_q
[i
+ 9] = dico2_isf
[ind
[1]][i
] * (1.0f
/ (1 << 15));
193 for (i
= 0; i
< 3; i
++)
194 isf_q
[i
] += dico21_isf
[ind
[2]][i
] * (1.0f
/ (1 << 15));
196 for (i
= 0; i
< 3; i
++)
197 isf_q
[i
+ 3] += dico22_isf
[ind
[3]][i
] * (1.0f
/ (1 << 15));
199 for (i
= 0; i
< 3; i
++)
200 isf_q
[i
+ 6] += dico23_isf
[ind
[4]][i
] * (1.0f
/ (1 << 15));
202 for (i
= 0; i
< 3; i
++)
203 isf_q
[i
+ 9] += dico24_isf
[ind
[5]][i
] * (1.0f
/ (1 << 15));
205 for (i
= 0; i
< 4; i
++)
206 isf_q
[i
+ 12] += dico25_isf
[ind
[6]][i
] * (1.0f
/ (1 << 15));
210 * Apply mean and past ISF values using the prediction factor.
211 * Updates past ISF vector.
213 * @param[in,out] isf_q Current quantized ISF
214 * @param[in,out] isf_past Past quantized ISF
217 static void isf_add_mean_and_past(float *isf_q
, float *isf_past
)
222 for (i
= 0; i
< LP_ORDER
; i
++) {
224 isf_q
[i
] += isf_mean
[i
] * (1.0f
/ (1 << 15));
225 isf_q
[i
] += PRED_FACTOR
* isf_past
[i
];
231 * Interpolate the fourth ISP vector from current and past frames
232 * to obtain an ISP vector for each subframe.
234 * @param[in,out] isp_q ISPs for each subframe
235 * @param[in] isp4_past Past ISP for subframe 4
237 static void interpolate_isp(double isp_q
[4][LP_ORDER
], const double *isp4_past
)
241 for (k
= 0; k
< 3; k
++) {
242 float c
= isfp_inter
[k
];
243 for (i
= 0; i
< LP_ORDER
; i
++)
244 isp_q
[k
][i
] = (1.0 - c
) * isp4_past
[i
] + c
* isp_q
[3][i
];
249 * Decode an adaptive codebook index into pitch lag (except 6k60, 8k85 modes).
250 * Calculate integer lag and fractional lag always using 1/4 resolution.
251 * In 1st and 3rd subframes the index is relative to last subframe integer lag.
253 * @param[out] lag_int Decoded integer pitch lag
254 * @param[out] lag_frac Decoded fractional pitch lag
255 * @param[in] pitch_index Adaptive codebook pitch index
256 * @param[in,out] base_lag_int Base integer lag used in relative subframes
257 * @param[in] subframe Current subframe index (0 to 3)
259 static void decode_pitch_lag_high(int *lag_int
, int *lag_frac
, int pitch_index
,
260 uint8_t *base_lag_int
, int subframe
)
262 if (subframe
== 0 || subframe
== 2) {
263 if (pitch_index
< 376) {
264 *lag_int
= (pitch_index
+ 137) >> 2;
265 *lag_frac
= pitch_index
- (*lag_int
<< 2) + 136;
266 } else if (pitch_index
< 440) {
267 *lag_int
= (pitch_index
+ 257 - 376) >> 1;
268 *lag_frac
= (pitch_index
- (*lag_int
<< 1) + 256 - 376) << 1;
269 /* the actual resolution is 1/2 but expressed as 1/4 */
271 *lag_int
= pitch_index
- 280;
274 /* minimum lag for next subframe */
275 *base_lag_int
= av_clip(*lag_int
- 8 - (*lag_frac
< 0),
276 AMRWB_P_DELAY_MIN
, AMRWB_P_DELAY_MAX
- 15);
277 // XXX: the spec states clearly that *base_lag_int should be
278 // the nearest integer to *lag_int (minus 8), but the ref code
279 // actually always uses its floor, I'm following the latter
281 *lag_int
= (pitch_index
+ 1) >> 2;
282 *lag_frac
= pitch_index
- (*lag_int
<< 2);
283 *lag_int
+= *base_lag_int
;
288 * Decode an adaptive codebook index into pitch lag for 8k85 and 6k60 modes.
289 * The description is analogous to decode_pitch_lag_high, but in 6k60 the
290 * relative index is used for all subframes except the first.
292 static void decode_pitch_lag_low(int *lag_int
, int *lag_frac
, int pitch_index
,
293 uint8_t *base_lag_int
, int subframe
, enum Mode mode
)
295 if (subframe
== 0 || (subframe
== 2 && mode
!= MODE_6k60
)) {
296 if (pitch_index
< 116) {
297 *lag_int
= (pitch_index
+ 69) >> 1;
298 *lag_frac
= (pitch_index
- (*lag_int
<< 1) + 68) << 1;
300 *lag_int
= pitch_index
- 24;
303 // XXX: same problem as before
304 *base_lag_int
= av_clip(*lag_int
- 8 - (*lag_frac
< 0),
305 AMRWB_P_DELAY_MIN
, AMRWB_P_DELAY_MAX
- 15);
307 *lag_int
= (pitch_index
+ 1) >> 1;
308 *lag_frac
= (pitch_index
- (*lag_int
<< 1)) << 1;
309 *lag_int
+= *base_lag_int
;
314 * Find the pitch vector by interpolating the past excitation at the
315 * pitch delay, which is obtained in this function.
317 * @param[in,out] ctx The context
318 * @param[in] amr_subframe Current subframe data
319 * @param[in] subframe Current subframe index (0 to 3)
321 static void decode_pitch_vector(AMRWBContext
*ctx
,
322 const AMRWBSubFrame
*amr_subframe
,
325 int pitch_lag_int
, pitch_lag_frac
;
327 float *exc
= ctx
->excitation
;
328 enum Mode mode
= ctx
->fr_cur_mode
;
330 if (mode
<= MODE_8k85
) {
331 decode_pitch_lag_low(&pitch_lag_int
, &pitch_lag_frac
, amr_subframe
->adap
,
332 &ctx
->base_pitch_lag
, subframe
, mode
);
334 decode_pitch_lag_high(&pitch_lag_int
, &pitch_lag_frac
, amr_subframe
->adap
,
335 &ctx
->base_pitch_lag
, subframe
);
337 ctx
->pitch_lag_int
= pitch_lag_int
;
338 pitch_lag_int
+= pitch_lag_frac
> 0;
340 /* Calculate the pitch vector by interpolating the past excitation at the
341 pitch lag using a hamming windowed sinc function */
342 ctx
->acelpf_ctx
.acelp_interpolatef(exc
,
343 exc
+ 1 - pitch_lag_int
,
345 pitch_lag_frac
+ (pitch_lag_frac
> 0 ? 0 : 4),
346 LP_ORDER
, AMRWB_SFR_SIZE
+ 1);
348 /* Check which pitch signal path should be used
349 * 6k60 and 8k85 modes have the ltp flag set to 0 */
350 if (amr_subframe
->ltp
) {
351 memcpy(ctx
->pitch_vector
, exc
, AMRWB_SFR_SIZE
* sizeof(float));
353 for (i
= 0; i
< AMRWB_SFR_SIZE
; i
++)
354 ctx
->pitch_vector
[i
] = 0.18 * exc
[i
- 1] + 0.64 * exc
[i
] +
356 memcpy(exc
, ctx
->pitch_vector
, AMRWB_SFR_SIZE
* sizeof(float));
360 /** Get x bits in the index interval [lsb,lsb+len-1] inclusive */
361 #define BIT_STR(x,lsb,len) (((x) >> (lsb)) & ((1 << (len)) - 1))
363 /** Get the bit at specified position */
364 #define BIT_POS(x, p) (((x) >> (p)) & 1)
367 * The next six functions decode_[i]p_track decode exactly i pulses
368 * positions and amplitudes (-1 or 1) in a subframe track using
369 * an encoded pulse indexing (TS 26.190 section 5.8.2).
371 * The results are given in out[], in which a negative number means
372 * amplitude -1 and vice versa (i.e., ampl(x) = x / abs(x) ).
374 * @param[out] out Output buffer (writes i elements)
375 * @param[in] code Pulse index (no. of bits varies, see below)
376 * @param[in] m (log2) Number of potential positions
377 * @param[in] off Offset for decoded positions
379 static inline void decode_1p_track(int *out
, int code
, int m
, int off
)
381 int pos
= BIT_STR(code
, 0, m
) + off
; ///code: m+1 bits
383 out
[0] = BIT_POS(code
, m
) ? -pos
: pos
;
386 static inline void decode_2p_track(int *out
, int code
, int m
, int off
) ///code: 2m+1 bits
388 int pos0
= BIT_STR(code
, m
, m
) + off
;
389 int pos1
= BIT_STR(code
, 0, m
) + off
;
391 out
[0] = BIT_POS(code
, 2*m
) ? -pos0
: pos0
;
392 out
[1] = BIT_POS(code
, 2*m
) ? -pos1
: pos1
;
393 out
[1] = pos0
> pos1
? -out
[1] : out
[1];
396 static void decode_3p_track(int *out
, int code
, int m
, int off
) ///code: 3m+1 bits
398 int half_2p
= BIT_POS(code
, 2*m
- 1) << (m
- 1);
400 decode_2p_track(out
, BIT_STR(code
, 0, 2*m
- 1),
401 m
- 1, off
+ half_2p
);
402 decode_1p_track(out
+ 2, BIT_STR(code
, 2*m
, m
+ 1), m
, off
);
405 static void decode_4p_track(int *out
, int code
, int m
, int off
) ///code: 4m bits
407 int half_4p
, subhalf_2p
;
408 int b_offset
= 1 << (m
- 1);
410 switch (BIT_STR(code
, 4*m
- 2, 2)) { /* case ID (2 bits) */
411 case 0: /* 0 pulses in A, 4 pulses in B or vice versa */
412 half_4p
= BIT_POS(code
, 4*m
- 3) << (m
- 1); // which has 4 pulses
413 subhalf_2p
= BIT_POS(code
, 2*m
- 3) << (m
- 2);
415 decode_2p_track(out
, BIT_STR(code
, 0, 2*m
- 3),
416 m
- 2, off
+ half_4p
+ subhalf_2p
);
417 decode_2p_track(out
+ 2, BIT_STR(code
, 2*m
- 2, 2*m
- 1),
418 m
- 1, off
+ half_4p
);
420 case 1: /* 1 pulse in A, 3 pulses in B */
421 decode_1p_track(out
, BIT_STR(code
, 3*m
- 2, m
),
423 decode_3p_track(out
+ 1, BIT_STR(code
, 0, 3*m
- 2),
424 m
- 1, off
+ b_offset
);
426 case 2: /* 2 pulses in each half */
427 decode_2p_track(out
, BIT_STR(code
, 2*m
- 1, 2*m
- 1),
429 decode_2p_track(out
+ 2, BIT_STR(code
, 0, 2*m
- 1),
430 m
- 1, off
+ b_offset
);
432 case 3: /* 3 pulses in A, 1 pulse in B */
433 decode_3p_track(out
, BIT_STR(code
, m
, 3*m
- 2),
435 decode_1p_track(out
+ 3, BIT_STR(code
, 0, m
),
436 m
- 1, off
+ b_offset
);
441 static void decode_5p_track(int *out
, int code
, int m
, int off
) ///code: 5m bits
443 int half_3p
= BIT_POS(code
, 5*m
- 1) << (m
- 1);
445 decode_3p_track(out
, BIT_STR(code
, 2*m
+ 1, 3*m
- 2),
446 m
- 1, off
+ half_3p
);
448 decode_2p_track(out
+ 3, BIT_STR(code
, 0, 2*m
+ 1), m
, off
);
451 static void decode_6p_track(int *out
, int code
, int m
, int off
) ///code: 6m-2 bits
453 int b_offset
= 1 << (m
- 1);
454 /* which half has more pulses in cases 0 to 2 */
455 int half_more
= BIT_POS(code
, 6*m
- 5) << (m
- 1);
456 int half_other
= b_offset
- half_more
;
458 switch (BIT_STR(code
, 6*m
- 4, 2)) { /* case ID (2 bits) */
459 case 0: /* 0 pulses in A, 6 pulses in B or vice versa */
460 decode_1p_track(out
, BIT_STR(code
, 0, m
),
461 m
- 1, off
+ half_more
);
462 decode_5p_track(out
+ 1, BIT_STR(code
, m
, 5*m
- 5),
463 m
- 1, off
+ half_more
);
465 case 1: /* 1 pulse in A, 5 pulses in B or vice versa */
466 decode_1p_track(out
, BIT_STR(code
, 0, m
),
467 m
- 1, off
+ half_other
);
468 decode_5p_track(out
+ 1, BIT_STR(code
, m
, 5*m
- 5),
469 m
- 1, off
+ half_more
);
471 case 2: /* 2 pulses in A, 4 pulses in B or vice versa */
472 decode_2p_track(out
, BIT_STR(code
, 0, 2*m
- 1),
473 m
- 1, off
+ half_other
);
474 decode_4p_track(out
+ 2, BIT_STR(code
, 2*m
- 1, 4*m
- 4),
475 m
- 1, off
+ half_more
);
477 case 3: /* 3 pulses in A, 3 pulses in B */
478 decode_3p_track(out
, BIT_STR(code
, 3*m
- 2, 3*m
- 2),
480 decode_3p_track(out
+ 3, BIT_STR(code
, 0, 3*m
- 2),
481 m
- 1, off
+ b_offset
);
487 * Decode the algebraic codebook index to pulse positions and signs,
488 * then construct the algebraic codebook vector.
490 * @param[out] fixed_vector Buffer for the fixed codebook excitation
491 * @param[in] pulse_hi MSBs part of the pulse index array (higher modes only)
492 * @param[in] pulse_lo LSBs part of the pulse index array
493 * @param[in] mode Mode of the current frame
495 static void decode_fixed_vector(float *fixed_vector
, const uint16_t *pulse_hi
,
496 const uint16_t *pulse_lo
, const enum Mode mode
)
498 /* sig_pos stores for each track the decoded pulse position indexes
499 * (1-based) multiplied by its corresponding amplitude (+1 or -1) */
501 int spacing
= (mode
== MODE_6k60
) ? 2 : 4;
506 for (i
= 0; i
< 2; i
++)
507 decode_1p_track(sig_pos
[i
], pulse_lo
[i
], 5, 1);
510 for (i
= 0; i
< 4; i
++)
511 decode_1p_track(sig_pos
[i
], pulse_lo
[i
], 4, 1);
514 for (i
= 0; i
< 4; i
++)
515 decode_2p_track(sig_pos
[i
], pulse_lo
[i
], 4, 1);
518 for (i
= 0; i
< 2; i
++)
519 decode_3p_track(sig_pos
[i
], pulse_lo
[i
], 4, 1);
520 for (i
= 2; i
< 4; i
++)
521 decode_2p_track(sig_pos
[i
], pulse_lo
[i
], 4, 1);
524 for (i
= 0; i
< 4; i
++)
525 decode_3p_track(sig_pos
[i
], pulse_lo
[i
], 4, 1);
528 for (i
= 0; i
< 4; i
++)
529 decode_4p_track(sig_pos
[i
], (int) pulse_lo
[i
] +
530 ((int) pulse_hi
[i
] << 14), 4, 1);
533 for (i
= 0; i
< 2; i
++)
534 decode_5p_track(sig_pos
[i
], (int) pulse_lo
[i
] +
535 ((int) pulse_hi
[i
] << 10), 4, 1);
536 for (i
= 2; i
< 4; i
++)
537 decode_4p_track(sig_pos
[i
], (int) pulse_lo
[i
] +
538 ((int) pulse_hi
[i
] << 14), 4, 1);
542 for (i
= 0; i
< 4; i
++)
543 decode_6p_track(sig_pos
[i
], (int) pulse_lo
[i
] +
544 ((int) pulse_hi
[i
] << 11), 4, 1);
548 memset(fixed_vector
, 0, sizeof(float) * AMRWB_SFR_SIZE
);
550 for (i
= 0; i
< 4; i
++)
551 for (j
= 0; j
< pulses_nb_per_mode_tr
[mode
][i
]; j
++) {
552 int pos
= (FFABS(sig_pos
[i
][j
]) - 1) * spacing
+ i
;
554 fixed_vector
[pos
] += sig_pos
[i
][j
] < 0 ? -1.0 : 1.0;
559 * Decode pitch gain and fixed gain correction factor.
561 * @param[in] vq_gain Vector-quantized index for gains
562 * @param[in] mode Mode of the current frame
563 * @param[out] fixed_gain_factor Decoded fixed gain correction factor
564 * @param[out] pitch_gain Decoded pitch gain
566 static void decode_gains(const uint8_t vq_gain
, const enum Mode mode
,
567 float *fixed_gain_factor
, float *pitch_gain
)
569 const int16_t *gains
= (mode
<= MODE_8k85
? qua_gain_6b
[vq_gain
] :
570 qua_gain_7b
[vq_gain
]);
572 *pitch_gain
= gains
[0] * (1.0f
/ (1 << 14));
573 *fixed_gain_factor
= gains
[1] * (1.0f
/ (1 << 11));
577 * Apply pitch sharpening filters to the fixed codebook vector.
579 * @param[in] ctx The context
580 * @param[in,out] fixed_vector Fixed codebook excitation
582 // XXX: Spec states this procedure should be applied when the pitch
583 // lag is less than 64, but this checking seems absent in reference and AMR-NB
584 static void pitch_sharpening(AMRWBContext
*ctx
, float *fixed_vector
)
589 for (i
= AMRWB_SFR_SIZE
- 1; i
!= 0; i
--)
590 fixed_vector
[i
] -= fixed_vector
[i
- 1] * ctx
->tilt_coef
;
592 /* Periodicity enhancement part */
593 for (i
= ctx
->pitch_lag_int
; i
< AMRWB_SFR_SIZE
; i
++)
594 fixed_vector
[i
] += fixed_vector
[i
- ctx
->pitch_lag_int
] * 0.85;
598 * Calculate the voicing factor (-1.0 = unvoiced to 1.0 = voiced).
600 * @param[in] p_vector, f_vector Pitch and fixed excitation vectors
601 * @param[in] p_gain, f_gain Pitch and fixed gains
602 * @param[in] ctx The context
604 // XXX: There is something wrong with the precision here! The magnitudes
605 // of the energies are not correct. Please check the reference code carefully
606 static float voice_factor(float *p_vector
, float p_gain
,
607 float *f_vector
, float f_gain
,
610 double p_ener
= (double) ctx
->dot_productf(p_vector
, p_vector
,
613 double f_ener
= (double) ctx
->dot_productf(f_vector
, f_vector
,
617 return (p_ener
- f_ener
) / (p_ener
+ f_ener
);
621 * Reduce fixed vector sparseness by smoothing with one of three IR filters,
622 * also known as "adaptive phase dispersion".
624 * @param[in] ctx The context
625 * @param[in,out] fixed_vector Unfiltered fixed vector
626 * @param[out] buf Space for modified vector if necessary
628 * @return The potentially overwritten filtered fixed vector address
630 static float *anti_sparseness(AMRWBContext
*ctx
,
631 float *fixed_vector
, float *buf
)
635 if (ctx
->fr_cur_mode
> MODE_8k85
) // no filtering in higher modes
638 if (ctx
->pitch_gain
[0] < 0.6) {
639 ir_filter_nr
= 0; // strong filtering
640 } else if (ctx
->pitch_gain
[0] < 0.9) {
641 ir_filter_nr
= 1; // medium filtering
643 ir_filter_nr
= 2; // no filtering
646 if (ctx
->fixed_gain
[0] > 3.0 * ctx
->fixed_gain
[1]) {
647 if (ir_filter_nr
< 2)
652 for (i
= 0; i
< 6; i
++)
653 if (ctx
->pitch_gain
[i
] < 0.6)
659 if (ir_filter_nr
> ctx
->prev_ir_filter_nr
+ 1)
663 /* update ir filter strength history */
664 ctx
->prev_ir_filter_nr
= ir_filter_nr
;
666 ir_filter_nr
+= (ctx
->fr_cur_mode
== MODE_8k85
);
668 if (ir_filter_nr
< 2) {
670 const float *coef
= ir_filters_lookup
[ir_filter_nr
];
672 /* Circular convolution code in the reference
673 * decoder was modified to avoid using one
674 * extra array. The filtered vector is given by:
676 * c2(n) = sum(i,0,len-1){ c(i) * coef( (n - i + len) % len ) }
679 memset(buf
, 0, sizeof(float) * AMRWB_SFR_SIZE
);
680 for (i
= 0; i
< AMRWB_SFR_SIZE
; i
++)
682 ff_celp_circ_addf(buf
, buf
, coef
, i
, fixed_vector
[i
],
691 * Calculate a stability factor {teta} based on distance between
692 * current and past isf. A value of 1 shows maximum signal stability.
694 static float stability_factor(const float *isf
, const float *isf_past
)
699 for (i
= 0; i
< LP_ORDER
- 1; i
++)
700 acc
+= (isf
[i
] - isf_past
[i
]) * (isf
[i
] - isf_past
[i
]);
702 // XXX: This part is not so clear from the reference code
703 // the result is more accurate changing the "/ 256" to "* 512"
704 return FFMAX(0.0, 1.25 - acc
* 0.8 * 512);
708 * Apply a non-linear fixed gain smoothing in order to reduce
709 * fluctuation in the energy of excitation.
711 * @param[in] fixed_gain Unsmoothed fixed gain
712 * @param[in,out] prev_tr_gain Previous threshold gain (updated)
713 * @param[in] voice_fac Frame voicing factor
714 * @param[in] stab_fac Frame stability factor
716 * @return The smoothed gain
718 static float noise_enhancer(float fixed_gain
, float *prev_tr_gain
,
719 float voice_fac
, float stab_fac
)
721 float sm_fac
= 0.5 * (1 - voice_fac
) * stab_fac
;
724 // XXX: the following fixed-point constants used to in(de)crement
725 // gain by 1.5dB were taken from the reference code, maybe it could
727 if (fixed_gain
< *prev_tr_gain
) {
728 g0
= FFMIN(*prev_tr_gain
, fixed_gain
+ fixed_gain
*
729 (6226 * (1.0f
/ (1 << 15)))); // +1.5 dB
731 g0
= FFMAX(*prev_tr_gain
, fixed_gain
*
732 (27536 * (1.0f
/ (1 << 15)))); // -1.5 dB
734 *prev_tr_gain
= g0
; // update next frame threshold
736 return sm_fac
* g0
+ (1 - sm_fac
) * fixed_gain
;
740 * Filter the fixed_vector to emphasize the higher frequencies.
742 * @param[in,out] fixed_vector Fixed codebook vector
743 * @param[in] voice_fac Frame voicing factor
745 static void pitch_enhancer(float *fixed_vector
, float voice_fac
)
748 float cpe
= 0.125 * (1 + voice_fac
);
749 float last
= fixed_vector
[0]; // holds c(i - 1)
751 fixed_vector
[0] -= cpe
* fixed_vector
[1];
753 for (i
= 1; i
< AMRWB_SFR_SIZE
- 1; i
++) {
754 float cur
= fixed_vector
[i
];
756 fixed_vector
[i
] -= cpe
* (last
+ fixed_vector
[i
+ 1]);
760 fixed_vector
[AMRWB_SFR_SIZE
- 1] -= cpe
* last
;
764 * Conduct 16th order linear predictive coding synthesis from excitation.
766 * @param[in] ctx Pointer to the AMRWBContext
767 * @param[in] lpc Pointer to the LPC coefficients
768 * @param[out] excitation Buffer for synthesis final excitation
769 * @param[in] fixed_gain Fixed codebook gain for synthesis
770 * @param[in] fixed_vector Algebraic codebook vector
771 * @param[in,out] samples Pointer to the output samples and memory
773 static void synthesis(AMRWBContext
*ctx
, float *lpc
, float *excitation
,
774 float fixed_gain
, const float *fixed_vector
,
777 ctx
->acelpv_ctx
.weighted_vector_sumf(excitation
, ctx
->pitch_vector
, fixed_vector
,
778 ctx
->pitch_gain
[0], fixed_gain
, AMRWB_SFR_SIZE
);
780 /* emphasize pitch vector contribution in low bitrate modes */
781 if (ctx
->pitch_gain
[0] > 0.5 && ctx
->fr_cur_mode
<= MODE_8k85
) {
783 float energy
= ctx
->celpm_ctx
.dot_productf(excitation
, excitation
,
786 // XXX: Weird part in both ref code and spec. A unknown parameter
787 // {beta} seems to be identical to the current pitch gain
788 float pitch_factor
= 0.25 * ctx
->pitch_gain
[0] * ctx
->pitch_gain
[0];
790 for (i
= 0; i
< AMRWB_SFR_SIZE
; i
++)
791 excitation
[i
] += pitch_factor
* ctx
->pitch_vector
[i
];
793 ff_scale_vector_to_given_sum_of_squares(excitation
, excitation
,
794 energy
, AMRWB_SFR_SIZE
);
797 ctx
->celpf_ctx
.celp_lp_synthesis_filterf(samples
, lpc
, excitation
,
798 AMRWB_SFR_SIZE
, LP_ORDER
);
802 * Apply to synthesis a de-emphasis filter of the form:
803 * H(z) = 1 / (1 - m * z^-1)
805 * @param[out] out Output buffer
806 * @param[in] in Input samples array with in[-1]
807 * @param[in] m Filter coefficient
808 * @param[in,out] mem State from last filtering
810 static void de_emphasis(float *out
, float *in
, float m
, float mem
[1])
814 out
[0] = in
[0] + m
* mem
[0];
816 for (i
= 1; i
< AMRWB_SFR_SIZE
; i
++)
817 out
[i
] = in
[i
] + out
[i
- 1] * m
;
819 mem
[0] = out
[AMRWB_SFR_SIZE
- 1];
823 * Upsample a signal by 5/4 ratio (from 12.8kHz to 16kHz) using
824 * a FIR interpolation filter. Uses past data from before *in address.
826 * @param[out] out Buffer for interpolated signal
827 * @param[in] in Current signal data (length 0.8*o_size)
828 * @param[in] o_size Output signal length
829 * @param[in] ctx The context
831 static void upsample_5_4(float *out
, const float *in
, int o_size
, CELPMContext
*ctx
)
833 const float *in0
= in
- UPS_FIR_SIZE
+ 1;
835 int int_part
= 0, frac_part
;
838 for (j
= 0; j
< o_size
/ 5; j
++) {
839 out
[i
] = in
[int_part
];
843 for (k
= 1; k
< 5; k
++) {
844 out
[i
] = ctx
->dot_productf(in0
+ int_part
,
845 upsample_fir
[4 - frac_part
],
855 * Calculate the high-band gain based on encoded index (23k85 mode) or
856 * on the low-band speech signal and the Voice Activity Detection flag.
858 * @param[in] ctx The context
859 * @param[in] synth LB speech synthesis at 12.8k
860 * @param[in] hb_idx Gain index for mode 23k85 only
861 * @param[in] vad VAD flag for the frame
863 static float find_hb_gain(AMRWBContext
*ctx
, const float *synth
,
864 uint16_t hb_idx
, uint8_t vad
)
869 if (ctx
->fr_cur_mode
== MODE_23k85
)
870 return qua_hb_gain
[hb_idx
] * (1.0f
/ (1 << 14));
872 tilt
= ctx
->celpm_ctx
.dot_productf(synth
, synth
+ 1, AMRWB_SFR_SIZE
- 1) /
873 ctx
->celpm_ctx
.dot_productf(synth
, synth
, AMRWB_SFR_SIZE
);
875 /* return gain bounded by [0.1, 1.0] */
876 return av_clipf((1.0 - FFMAX(0.0, tilt
)) * (1.25 - 0.25 * wsp
), 0.1, 1.0);
880 * Generate the high-band excitation with the same energy from the lower
881 * one and scaled by the given gain.
883 * @param[in] ctx The context
884 * @param[out] hb_exc Buffer for the excitation
885 * @param[in] synth_exc Low-band excitation used for synthesis
886 * @param[in] hb_gain Wanted excitation gain
888 static void scaled_hb_excitation(AMRWBContext
*ctx
, float *hb_exc
,
889 const float *synth_exc
, float hb_gain
)
892 float energy
= ctx
->celpm_ctx
.dot_productf(synth_exc
, synth_exc
,
895 /* Generate a white-noise excitation */
896 for (i
= 0; i
< AMRWB_SFR_SIZE_16k
; i
++)
897 hb_exc
[i
] = 32768.0 - (uint16_t) av_lfg_get(&ctx
->prng
);
899 ff_scale_vector_to_given_sum_of_squares(hb_exc
, hb_exc
,
900 energy
* hb_gain
* hb_gain
,
905 * Calculate the auto-correlation for the ISF difference vector.
907 static float auto_correlation(float *diff_isf
, float mean
, int lag
)
912 for (i
= 7; i
< LP_ORDER
- 2; i
++) {
913 float prod
= (diff_isf
[i
] - mean
) * (diff_isf
[i
- lag
] - mean
);
920 * Extrapolate a ISF vector to the 16kHz range (20th order LP)
921 * used at mode 6k60 LP filter for the high frequency band.
923 * @param[out] isf Buffer for extrapolated isf; contains LP_ORDER
926 static void extrapolate_isf(float isf
[LP_ORDER_16k
])
928 float diff_isf
[LP_ORDER
- 2], diff_mean
;
931 int i
, j
, i_max_corr
;
933 isf
[LP_ORDER_16k
- 1] = isf
[LP_ORDER
- 1];
935 /* Calculate the difference vector */
936 for (i
= 0; i
< LP_ORDER
- 2; i
++)
937 diff_isf
[i
] = isf
[i
+ 1] - isf
[i
];
940 for (i
= 2; i
< LP_ORDER
- 2; i
++)
941 diff_mean
+= diff_isf
[i
] * (1.0f
/ (LP_ORDER
- 4));
943 /* Find which is the maximum autocorrelation */
945 for (i
= 0; i
< 3; i
++) {
946 corr_lag
[i
] = auto_correlation(diff_isf
, diff_mean
, i
+ 2);
948 if (corr_lag
[i
] > corr_lag
[i_max_corr
])
953 for (i
= LP_ORDER
- 1; i
< LP_ORDER_16k
- 1; i
++)
954 isf
[i
] = isf
[i
- 1] + isf
[i
- 1 - i_max_corr
]
955 - isf
[i
- 2 - i_max_corr
];
957 /* Calculate an estimate for ISF(18) and scale ISF based on the error */
958 est
= 7965 + (isf
[2] - isf
[3] - isf
[4]) / 6.0;
959 scale
= 0.5 * (FFMIN(est
, 7600) - isf
[LP_ORDER
- 2]) /
960 (isf
[LP_ORDER_16k
- 2] - isf
[LP_ORDER
- 2]);
962 for (i
= LP_ORDER
- 1, j
= 0; i
< LP_ORDER_16k
- 1; i
++, j
++)
963 diff_isf
[j
] = scale
* (isf
[i
] - isf
[i
- 1]);
965 /* Stability insurance */
966 for (i
= 1; i
< LP_ORDER_16k
- LP_ORDER
; i
++)
967 if (diff_isf
[i
] + diff_isf
[i
- 1] < 5.0) {
968 if (diff_isf
[i
] > diff_isf
[i
- 1]) {
969 diff_isf
[i
- 1] = 5.0 - diff_isf
[i
];
971 diff_isf
[i
] = 5.0 - diff_isf
[i
- 1];
974 for (i
= LP_ORDER
- 1, j
= 0; i
< LP_ORDER_16k
- 1; i
++, j
++)
975 isf
[i
] = isf
[i
- 1] + diff_isf
[j
] * (1.0f
/ (1 << 15));
977 /* Scale the ISF vector for 16000 Hz */
978 for (i
= 0; i
< LP_ORDER_16k
- 1; i
++)
983 * Spectral expand the LP coefficients using the equation:
984 * y[i] = x[i] * (gamma ** i)
986 * @param[out] out Output buffer (may use input array)
987 * @param[in] lpc LP coefficients array
988 * @param[in] gamma Weighting factor
989 * @param[in] size LP array size
991 static void lpc_weighting(float *out
, const float *lpc
, float gamma
, int size
)
996 for (i
= 0; i
< size
; i
++) {
997 out
[i
] = lpc
[i
] * fac
;
1003 * Conduct 20th order linear predictive coding synthesis for the high
1004 * frequency band excitation at 16kHz.
1006 * @param[in] ctx The context
1007 * @param[in] subframe Current subframe index (0 to 3)
1008 * @param[in,out] samples Pointer to the output speech samples
1009 * @param[in] exc Generated white-noise scaled excitation
1010 * @param[in] isf Current frame isf vector
1011 * @param[in] isf_past Past frame final isf vector
1013 static void hb_synthesis(AMRWBContext
*ctx
, int subframe
, float *samples
,
1014 const float *exc
, const float *isf
, const float *isf_past
)
1016 float hb_lpc
[LP_ORDER_16k
];
1017 enum Mode mode
= ctx
->fr_cur_mode
;
1019 if (mode
== MODE_6k60
) {
1020 float e_isf
[LP_ORDER_16k
]; // ISF vector for extrapolation
1021 double e_isp
[LP_ORDER_16k
];
1023 ctx
->acelpv_ctx
.weighted_vector_sumf(e_isf
, isf_past
, isf
, isfp_inter
[subframe
],
1024 1.0 - isfp_inter
[subframe
], LP_ORDER
);
1026 extrapolate_isf(e_isf
);
1028 e_isf
[LP_ORDER_16k
- 1] *= 2.0;
1029 ff_acelp_lsf2lspd(e_isp
, e_isf
, LP_ORDER_16k
);
1030 ff_amrwb_lsp2lpc(e_isp
, hb_lpc
, LP_ORDER_16k
);
1032 lpc_weighting(hb_lpc
, hb_lpc
, 0.9, LP_ORDER_16k
);
1034 lpc_weighting(hb_lpc
, ctx
->lp_coef
[subframe
], 0.6, LP_ORDER
);
1037 ctx
->celpf_ctx
.celp_lp_synthesis_filterf(samples
, hb_lpc
, exc
, AMRWB_SFR_SIZE_16k
,
1038 (mode
== MODE_6k60
) ? LP_ORDER_16k
: LP_ORDER
);
1042 * Apply a 15th order filter to high-band samples.
1043 * The filter characteristic depends on the given coefficients.
1045 * @param[out] out Buffer for filtered output
1046 * @param[in] fir_coef Filter coefficients
1047 * @param[in,out] mem State from last filtering (updated)
1048 * @param[in] in Input speech data (high-band)
1050 * @remark It is safe to pass the same array in in and out parameters
1053 #ifndef hb_fir_filter
1054 static void hb_fir_filter(float *out
, const float fir_coef
[HB_FIR_SIZE
+ 1],
1055 float mem
[HB_FIR_SIZE
], const float *in
)
1058 float data
[AMRWB_SFR_SIZE_16k
+ HB_FIR_SIZE
]; // past and current samples
1060 memcpy(data
, mem
, HB_FIR_SIZE
* sizeof(float));
1061 memcpy(data
+ HB_FIR_SIZE
, in
, AMRWB_SFR_SIZE_16k
* sizeof(float));
1063 for (i
= 0; i
< AMRWB_SFR_SIZE_16k
; i
++) {
1065 for (j
= 0; j
<= HB_FIR_SIZE
; j
++)
1066 out
[i
] += data
[i
+ j
] * fir_coef
[j
];
1069 memcpy(mem
, data
+ AMRWB_SFR_SIZE_16k
, HB_FIR_SIZE
* sizeof(float));
1071 #endif /* hb_fir_filter */
1074 * Update context state before the next subframe.
1076 static void update_sub_state(AMRWBContext
*ctx
)
1078 memmove(&ctx
->excitation_buf
[0], &ctx
->excitation_buf
[AMRWB_SFR_SIZE
],
1079 (AMRWB_P_DELAY_MAX
+ LP_ORDER
+ 1) * sizeof(float));
1081 memmove(&ctx
->pitch_gain
[1], &ctx
->pitch_gain
[0], 5 * sizeof(float));
1082 memmove(&ctx
->fixed_gain
[1], &ctx
->fixed_gain
[0], 1 * sizeof(float));
1084 memmove(&ctx
->samples_az
[0], &ctx
->samples_az
[AMRWB_SFR_SIZE
],
1085 LP_ORDER
* sizeof(float));
1086 memmove(&ctx
->samples_up
[0], &ctx
->samples_up
[AMRWB_SFR_SIZE
],
1087 UPS_MEM_SIZE
* sizeof(float));
1088 memmove(&ctx
->samples_hb
[0], &ctx
->samples_hb
[AMRWB_SFR_SIZE_16k
],
1089 LP_ORDER_16k
* sizeof(float));
1092 static int amrwb_decode_frame(AVCodecContext
*avctx
, void *data
,
1093 int *got_frame_ptr
, AVPacket
*avpkt
)
1095 AMRWBContext
*ctx
= avctx
->priv_data
;
1096 AVFrame
*frame
= data
;
1097 AMRWBFrame
*cf
= &ctx
->frame
;
1098 const uint8_t *buf
= avpkt
->data
;
1099 int buf_size
= avpkt
->size
;
1100 int expected_fr_size
, header_size
;
1102 float spare_vector
[AMRWB_SFR_SIZE
]; // extra stack space to hold result from anti-sparseness processing
1103 float fixed_gain_factor
; // fixed gain correction factor (gamma)
1104 float *synth_fixed_vector
; // pointer to the fixed vector that synthesis should use
1105 float synth_fixed_gain
; // the fixed gain that synthesis should use
1106 float voice_fac
, stab_fac
; // parameters used for gain smoothing
1107 float synth_exc
[AMRWB_SFR_SIZE
]; // post-processed excitation for synthesis
1108 float hb_exc
[AMRWB_SFR_SIZE_16k
]; // excitation for the high frequency band
1109 float hb_samples
[AMRWB_SFR_SIZE_16k
]; // filtered high-band samples from synthesis
1113 /* get output buffer */
1114 frame
->nb_samples
= 4 * AMRWB_SFR_SIZE_16k
;
1115 if ((ret
= ff_get_buffer(avctx
, frame
, 0)) < 0)
1117 buf_out
= (float *)frame
->data
[0];
1119 header_size
= decode_mime_header(ctx
, buf
);
1120 if (ctx
->fr_cur_mode
> MODE_SID
) {
1121 av_log(avctx
, AV_LOG_ERROR
,
1122 "Invalid mode %d\n", ctx
->fr_cur_mode
);
1123 return AVERROR_INVALIDDATA
;
1125 expected_fr_size
= ((cf_sizes_wb
[ctx
->fr_cur_mode
] + 7) >> 3) + 1;
1127 if (buf_size
< expected_fr_size
) {
1128 av_log(avctx
, AV_LOG_ERROR
,
1129 "Frame too small (%d bytes). Truncated file?\n", buf_size
);
1131 return AVERROR_INVALIDDATA
;
1134 if (!ctx
->fr_quality
|| ctx
->fr_cur_mode
> MODE_SID
)
1135 av_log(avctx
, AV_LOG_ERROR
, "Encountered a bad or corrupted frame\n");
1137 if (ctx
->fr_cur_mode
== MODE_SID
) { /* Comfort noise frame */
1138 avpriv_request_sample(avctx
, "SID mode");
1139 return AVERROR_PATCHWELCOME
;
1142 ff_amr_bit_reorder((uint16_t *) &ctx
->frame
, sizeof(AMRWBFrame
),
1143 buf
+ header_size
, amr_bit_orderings_by_mode
[ctx
->fr_cur_mode
]);
1145 /* Decode the quantized ISF vector */
1146 if (ctx
->fr_cur_mode
== MODE_6k60
) {
1147 decode_isf_indices_36b(cf
->isp_id
, ctx
->isf_cur
);
1149 decode_isf_indices_46b(cf
->isp_id
, ctx
->isf_cur
);
1152 isf_add_mean_and_past(ctx
->isf_cur
, ctx
->isf_q_past
);
1153 ff_set_min_dist_lsf(ctx
->isf_cur
, MIN_ISF_SPACING
, LP_ORDER
- 1);
1155 stab_fac
= stability_factor(ctx
->isf_cur
, ctx
->isf_past_final
);
1157 ctx
->isf_cur
[LP_ORDER
- 1] *= 2.0;
1158 ff_acelp_lsf2lspd(ctx
->isp
[3], ctx
->isf_cur
, LP_ORDER
);
1160 /* Generate a ISP vector for each subframe */
1161 if (ctx
->first_frame
) {
1162 ctx
->first_frame
= 0;
1163 memcpy(ctx
->isp_sub4_past
, ctx
->isp
[3], LP_ORDER
* sizeof(double));
1165 interpolate_isp(ctx
->isp
, ctx
->isp_sub4_past
);
1167 for (sub
= 0; sub
< 4; sub
++)
1168 ff_amrwb_lsp2lpc(ctx
->isp
[sub
], ctx
->lp_coef
[sub
], LP_ORDER
);
1170 for (sub
= 0; sub
< 4; sub
++) {
1171 const AMRWBSubFrame
*cur_subframe
= &cf
->subframe
[sub
];
1172 float *sub_buf
= buf_out
+ sub
* AMRWB_SFR_SIZE_16k
;
1174 /* Decode adaptive codebook (pitch vector) */
1175 decode_pitch_vector(ctx
, cur_subframe
, sub
);
1176 /* Decode innovative codebook (fixed vector) */
1177 decode_fixed_vector(ctx
->fixed_vector
, cur_subframe
->pul_ih
,
1178 cur_subframe
->pul_il
, ctx
->fr_cur_mode
);
1180 pitch_sharpening(ctx
, ctx
->fixed_vector
);
1182 decode_gains(cur_subframe
->vq_gain
, ctx
->fr_cur_mode
,
1183 &fixed_gain_factor
, &ctx
->pitch_gain
[0]);
1185 ctx
->fixed_gain
[0] =
1186 ff_amr_set_fixed_gain(fixed_gain_factor
,
1187 ctx
->celpm_ctx
.dot_productf(ctx
->fixed_vector
,
1191 ctx
->prediction_error
,
1192 ENERGY_MEAN
, energy_pred_fac
);
1194 /* Calculate voice factor and store tilt for next subframe */
1195 voice_fac
= voice_factor(ctx
->pitch_vector
, ctx
->pitch_gain
[0],
1196 ctx
->fixed_vector
, ctx
->fixed_gain
[0],
1198 ctx
->tilt_coef
= voice_fac
* 0.25 + 0.25;
1200 /* Construct current excitation */
1201 for (i
= 0; i
< AMRWB_SFR_SIZE
; i
++) {
1202 ctx
->excitation
[i
] *= ctx
->pitch_gain
[0];
1203 ctx
->excitation
[i
] += ctx
->fixed_gain
[0] * ctx
->fixed_vector
[i
];
1204 ctx
->excitation
[i
] = truncf(ctx
->excitation
[i
]);
1207 /* Post-processing of excitation elements */
1208 synth_fixed_gain
= noise_enhancer(ctx
->fixed_gain
[0], &ctx
->prev_tr_gain
,
1209 voice_fac
, stab_fac
);
1211 synth_fixed_vector
= anti_sparseness(ctx
, ctx
->fixed_vector
,
1214 pitch_enhancer(synth_fixed_vector
, voice_fac
);
1216 synthesis(ctx
, ctx
->lp_coef
[sub
], synth_exc
, synth_fixed_gain
,
1217 synth_fixed_vector
, &ctx
->samples_az
[LP_ORDER
]);
1219 /* Synthesis speech post-processing */
1220 de_emphasis(&ctx
->samples_up
[UPS_MEM_SIZE
],
1221 &ctx
->samples_az
[LP_ORDER
], PREEMPH_FAC
, ctx
->demph_mem
);
1223 ctx
->acelpf_ctx
.acelp_apply_order_2_transfer_function(&ctx
->samples_up
[UPS_MEM_SIZE
],
1224 &ctx
->samples_up
[UPS_MEM_SIZE
], hpf_zeros
, hpf_31_poles
,
1225 hpf_31_gain
, ctx
->hpf_31_mem
, AMRWB_SFR_SIZE
);
1227 upsample_5_4(sub_buf
, &ctx
->samples_up
[UPS_FIR_SIZE
],
1228 AMRWB_SFR_SIZE_16k
, &ctx
->celpm_ctx
);
1230 /* High frequency band (6.4 - 7.0 kHz) generation part */
1231 ctx
->acelpf_ctx
.acelp_apply_order_2_transfer_function(hb_samples
,
1232 &ctx
->samples_up
[UPS_MEM_SIZE
], hpf_zeros
, hpf_400_poles
,
1233 hpf_400_gain
, ctx
->hpf_400_mem
, AMRWB_SFR_SIZE
);
1235 hb_gain
= find_hb_gain(ctx
, hb_samples
,
1236 cur_subframe
->hb_gain
, cf
->vad
);
1238 scaled_hb_excitation(ctx
, hb_exc
, synth_exc
, hb_gain
);
1240 hb_synthesis(ctx
, sub
, &ctx
->samples_hb
[LP_ORDER_16k
],
1241 hb_exc
, ctx
->isf_cur
, ctx
->isf_past_final
);
1243 /* High-band post-processing filters */
1244 hb_fir_filter(hb_samples
, bpf_6_7_coef
, ctx
->bpf_6_7_mem
,
1245 &ctx
->samples_hb
[LP_ORDER_16k
]);
1247 if (ctx
->fr_cur_mode
== MODE_23k85
)
1248 hb_fir_filter(hb_samples
, lpf_7_coef
, ctx
->lpf_7_mem
,
1251 /* Add the low and high frequency bands */
1252 for (i
= 0; i
< AMRWB_SFR_SIZE_16k
; i
++)
1253 sub_buf
[i
] = (sub_buf
[i
] + hb_samples
[i
]) * (1.0f
/ (1 << 15));
1255 /* Update buffers and history */
1256 update_sub_state(ctx
);
1259 /* update state for next frame */
1260 memcpy(ctx
->isp_sub4_past
, ctx
->isp
[3], LP_ORDER
* sizeof(ctx
->isp
[3][0]));
1261 memcpy(ctx
->isf_past_final
, ctx
->isf_cur
, LP_ORDER
* sizeof(float));
1265 return expected_fr_size
;
1268 AVCodec ff_amrwb_decoder
= {
1270 .long_name
= NULL_IF_CONFIG_SMALL("AMR-WB (Adaptive Multi-Rate WideBand)"),
1271 .type
= AVMEDIA_TYPE_AUDIO
,
1272 .id
= AV_CODEC_ID_AMR_WB
,
1273 .priv_data_size
= sizeof(AMRWBContext
),
1274 .init
= amrwb_decode_init
,
1275 .decode
= amrwb_decode_frame
,
1276 .capabilities
= CODEC_CAP_DR1
,
1277 .sample_fmts
= (const enum AVSampleFormat
[]){ AV_SAMPLE_FMT_FLT
,
1278 AV_SAMPLE_FMT_NONE
},