GCC Code Coverage Report
Directory: ../../../ffmpeg/ Exec Total Coverage
File: src/libavcodec/amrwbdec.c Lines: 492 508 96.9 %
Date: 2019-11-18 18:00:01 Branches: 224 239 93.7 %

Line Branch Exec Source
1
/*
2
 * AMR wideband decoder
3
 * Copyright (c) 2010 Marcelo Galvao Povoa
4
 *
5
 * This file is part of FFmpeg.
6
 *
7
 * FFmpeg is free software; you can redistribute it and/or
8
 * modify it under the terms of the GNU Lesser General Public
9
 * License as published by the Free Software Foundation; either
10
 * version 2.1 of the License, or (at your option) any later version.
11
 *
12
 * FFmpeg is distributed in the hope that it will be useful,
13
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14
 * MERCHANTABILITY or FITNESS FOR A particular PURPOSE.  See the GNU
15
 * Lesser General Public License for more details.
16
 *
17
 * You should have received a copy of the GNU Lesser General Public
18
 * License along with FFmpeg; if not, write to the Free Software
19
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20
 */
21
22
/**
23
 * @file
24
 * AMR wideband decoder
25
 */
26
27
#include "libavutil/channel_layout.h"
28
#include "libavutil/common.h"
29
#include "libavutil/float_dsp.h"
30
#include "libavutil/lfg.h"
31
32
#include "avcodec.h"
33
#include "lsp.h"
34
#include "celp_filters.h"
35
#include "celp_math.h"
36
#include "acelp_filters.h"
37
#include "acelp_vectors.h"
38
#include "acelp_pitch_delay.h"
39
#include "internal.h"
40
41
#define AMR_USE_16BIT_TABLES
42
#include "amr.h"
43
44
#include "amrwbdata.h"
45
#include "mips/amrwbdec_mips.h"
46
47
typedef struct AMRWBContext {
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
56
57
    float                   lp_coef[4][LP_ORDER]; ///< Linear Prediction Coefficients from ISP vector
58
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
61
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[]
64
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
67
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
71
72
    float                              tilt_coef; ///< {beta_1} related to the voicing of the previous subframe
73
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
77
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
81
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
86
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
93
94
} AMRWBContext;
95
96
20
static av_cold int amrwb_decode_init(AVCodecContext *avctx)
97
{
98
20
    AMRWBContext *ctx = avctx->priv_data;
99
    int i;
100
101
20
    if (avctx->channels > 1) {
102
        avpriv_report_missing_feature(avctx, "multi-channel AMR");
103
        return AVERROR_PATCHWELCOME;
104
    }
105
106
20
    avctx->channels       = 1;
107
20
    avctx->channel_layout = AV_CH_LAYOUT_MONO;
108
20
    if (!avctx->sample_rate)
109
        avctx->sample_rate = 16000;
110
20
    avctx->sample_fmt     = AV_SAMPLE_FMT_FLT;
111
112
20
    av_lfg_init(&ctx->prng, 1);
113
114
20
    ctx->excitation  = &ctx->excitation_buf[AMRWB_P_DELAY_MAX + LP_ORDER + 1];
115
20
    ctx->first_frame = 1;
116
117
340
    for (i = 0; i < LP_ORDER; i++)
118
320
        ctx->isf_past_final[i] = isf_init[i] * (1.0f / (1 << 15));
119
120
100
    for (i = 0; i < 4; i++)
121
80
        ctx->prediction_error[i] = MIN_ENERGY;
122
123
20
    ff_acelp_filter_init(&ctx->acelpf_ctx);
124
20
    ff_acelp_vectors_init(&ctx->acelpv_ctx);
125
20
    ff_celp_filter_init(&ctx->celpf_ctx);
126
20
    ff_celp_math_init(&ctx->celpm_ctx);
127
128
20
    return 0;
129
}
130
131
/**
132
 * Decode the frame header in the "MIME/storage" format. This format
133
 * is simpler and does not carry the auxiliary frame information.
134
 *
135
 * @param[in] ctx                  The Context
136
 * @param[in] buf                  Pointer to the input buffer
137
 *
138
 * @return The decoded header length in bytes
139
 */
140
6256
static int decode_mime_header(AMRWBContext *ctx, const uint8_t *buf)
141
{
142
    /* Decode frame header (1st octet) */
143
6256
    ctx->fr_cur_mode  = buf[0] >> 3 & 0x0F;
144
6256
    ctx->fr_quality   = (buf[0] & 0x4) == 0x4;
145
146
6256
    return 1;
147
}
148
149
/**
150
 * Decode quantized ISF vectors using 36-bit indexes (6K60 mode only).
151
 *
152
 * @param[in]  ind                 Array of 5 indexes
153
 * @param[out] isf_q               Buffer for isf_q[LP_ORDER]
154
 */
155
512
static void decode_isf_indices_36b(uint16_t *ind, float *isf_q)
156
{
157
    int i;
158
159
5120
    for (i = 0; i < 9; i++)
160
4608
        isf_q[i]      = dico1_isf[ind[0]][i]      * (1.0f / (1 << 15));
161
162
4096
    for (i = 0; i < 7; i++)
163
3584
        isf_q[i + 9]  = dico2_isf[ind[1]][i]      * (1.0f / (1 << 15));
164
165
3072
    for (i = 0; i < 5; i++)
166
2560
        isf_q[i]     += dico21_isf_36b[ind[2]][i] * (1.0f / (1 << 15));
167
168
2560
    for (i = 0; i < 4; i++)
169
2048
        isf_q[i + 5] += dico22_isf_36b[ind[3]][i] * (1.0f / (1 << 15));
170
171
4096
    for (i = 0; i < 7; i++)
172
3584
        isf_q[i + 9] += dico23_isf_36b[ind[4]][i] * (1.0f / (1 << 15));
173
512
}
174
175
/**
176
 * Decode quantized ISF vectors using 46-bit indexes (except 6K60 mode).
177
 *
178
 * @param[in]  ind                 Array of 7 indexes
179
 * @param[out] isf_q               Buffer for isf_q[LP_ORDER]
180
 */
181
5744
static void decode_isf_indices_46b(uint16_t *ind, float *isf_q)
182
{
183
    int i;
184
185
57440
    for (i = 0; i < 9; i++)
186
51696
        isf_q[i]       = dico1_isf[ind[0]][i]  * (1.0f / (1 << 15));
187
188
45952
    for (i = 0; i < 7; i++)
189
40208
        isf_q[i + 9]   = dico2_isf[ind[1]][i]  * (1.0f / (1 << 15));
190
191
22976
    for (i = 0; i < 3; i++)
192
17232
        isf_q[i]      += dico21_isf[ind[2]][i] * (1.0f / (1 << 15));
193
194
22976
    for (i = 0; i < 3; i++)
195
17232
        isf_q[i + 3]  += dico22_isf[ind[3]][i] * (1.0f / (1 << 15));
196
197
22976
    for (i = 0; i < 3; i++)
198
17232
        isf_q[i + 6]  += dico23_isf[ind[4]][i] * (1.0f / (1 << 15));
199
200
22976
    for (i = 0; i < 3; i++)
201
17232
        isf_q[i + 9]  += dico24_isf[ind[5]][i] * (1.0f / (1 << 15));
202
203
28720
    for (i = 0; i < 4; i++)
204
22976
        isf_q[i + 12] += dico25_isf[ind[6]][i] * (1.0f / (1 << 15));
205
5744
}
206
207
/**
208
 * Apply mean and past ISF values using the prediction factor.
209
 * Updates past ISF vector.
210
 *
211
 * @param[in,out] isf_q            Current quantized ISF
212
 * @param[in,out] isf_past         Past quantized ISF
213
 */
214
6256
static void isf_add_mean_and_past(float *isf_q, float *isf_past)
215
{
216
    int i;
217
    float tmp;
218
219
106352
    for (i = 0; i < LP_ORDER; i++) {
220
100096
        tmp = isf_q[i];
221
100096
        isf_q[i] += isf_mean[i] * (1.0f / (1 << 15));
222
100096
        isf_q[i] += PRED_FACTOR * isf_past[i];
223
100096
        isf_past[i] = tmp;
224
    }
225
6256
}
226
227
/**
228
 * Interpolate the fourth ISP vector from current and past frames
229
 * to obtain an ISP vector for each subframe.
230
 *
231
 * @param[in,out] isp_q            ISPs for each subframe
232
 * @param[in]     isp4_past        Past ISP for subframe 4
233
 */
234
6256
static void interpolate_isp(double isp_q[4][LP_ORDER], const double *isp4_past)
235
{
236
    int i, k;
237
238
25024
    for (k = 0; k < 3; k++) {
239
18768
        float c = isfp_inter[k];
240
319056
        for (i = 0; i < LP_ORDER; i++)
241
300288
            isp_q[k][i] = (1.0 - c) * isp4_past[i] + c * isp_q[3][i];
242
    }
243
6256
}
244
245
/**
246
 * Decode an adaptive codebook index into pitch lag (except 6k60, 8k85 modes).
247
 * Calculate integer lag and fractional lag always using 1/4 resolution.
248
 * In 1st and 3rd subframes the index is relative to last subframe integer lag.
249
 *
250
 * @param[out]    lag_int          Decoded integer pitch lag
251
 * @param[out]    lag_frac         Decoded fractional pitch lag
252
 * @param[in]     pitch_index      Adaptive codebook pitch index
253
 * @param[in,out] base_lag_int     Base integer lag used in relative subframes
254
 * @param[in]     subframe         Current subframe index (0 to 3)
255
 */
256
20928
static void decode_pitch_lag_high(int *lag_int, int *lag_frac, int pitch_index,
257
                                  uint8_t *base_lag_int, int subframe)
258
{
259

20928
    if (subframe == 0 || subframe == 2) {
260
10464
        if (pitch_index < 376) {
261
7610
            *lag_int  = (pitch_index + 137) >> 2;
262
7610
            *lag_frac = pitch_index - (*lag_int << 2) + 136;
263
2854
        } else if (pitch_index < 440) {
264
1248
            *lag_int  = (pitch_index + 257 - 376) >> 1;
265
1248
            *lag_frac = (pitch_index - (*lag_int << 1) + 256 - 376) * 2;
266
            /* the actual resolution is 1/2 but expressed as 1/4 */
267
        } else {
268
1606
            *lag_int  = pitch_index - 280;
269
1606
            *lag_frac = 0;
270
        }
271
        /* minimum lag for next subframe */
272
10464
        *base_lag_int = av_clip(*lag_int - 8 - (*lag_frac < 0),
273
                                AMRWB_P_DELAY_MIN, AMRWB_P_DELAY_MAX - 15);
274
        // XXX: the spec states clearly that *base_lag_int should be
275
        // the nearest integer to *lag_int (minus 8), but the ref code
276
        // actually always uses its floor, I'm following the latter
277
    } else {
278
10464
        *lag_int  = (pitch_index + 1) >> 2;
279
10464
        *lag_frac = pitch_index - (*lag_int << 2);
280
10464
        *lag_int += *base_lag_int;
281
    }
282
20928
}
283
284
/**
285
 * Decode an adaptive codebook index into pitch lag for 8k85 and 6k60 modes.
286
 * The description is analogous to decode_pitch_lag_high, but in 6k60 the
287
 * relative index is used for all subframes except the first.
288
 */
289
4096
static void decode_pitch_lag_low(int *lag_int, int *lag_frac, int pitch_index,
290
                                 uint8_t *base_lag_int, int subframe, enum Mode mode)
291
{
292

4096
    if (subframe == 0 || (subframe == 2 && mode != MODE_6k60)) {
293
1536
        if (pitch_index < 116) {
294
847
            *lag_int  = (pitch_index + 69) >> 1;
295
847
            *lag_frac = (pitch_index - (*lag_int << 1) + 68) * 2;
296
        } else {
297
689
            *lag_int  = pitch_index - 24;
298
689
            *lag_frac = 0;
299
        }
300
        // XXX: same problem as before
301
1536
        *base_lag_int = av_clip(*lag_int - 8 - (*lag_frac < 0),
302
                                AMRWB_P_DELAY_MIN, AMRWB_P_DELAY_MAX - 15);
303
    } else {
304
2560
        *lag_int  = (pitch_index + 1) >> 1;
305
2560
        *lag_frac = (pitch_index - (*lag_int << 1)) * 2;
306
2560
        *lag_int += *base_lag_int;
307
    }
308
4096
}
309
310
/**
311
 * Find the pitch vector by interpolating the past excitation at the
312
 * pitch delay, which is obtained in this function.
313
 *
314
 * @param[in,out] ctx              The context
315
 * @param[in]     amr_subframe     Current subframe data
316
 * @param[in]     subframe         Current subframe index (0 to 3)
317
 */
318
25024
static void decode_pitch_vector(AMRWBContext *ctx,
319
                                const AMRWBSubFrame *amr_subframe,
320
                                const int subframe)
321
{
322
    int pitch_lag_int, pitch_lag_frac;
323
    int i;
324
25024
    float *exc     = ctx->excitation;
325
25024
    enum Mode mode = ctx->fr_cur_mode;
326
327
25024
    if (mode <= MODE_8k85) {
328
4096
        decode_pitch_lag_low(&pitch_lag_int, &pitch_lag_frac, amr_subframe->adap,
329
                              &ctx->base_pitch_lag, subframe, mode);
330
    } else
331
20928
        decode_pitch_lag_high(&pitch_lag_int, &pitch_lag_frac, amr_subframe->adap,
332
                              &ctx->base_pitch_lag, subframe);
333
334
25024
    ctx->pitch_lag_int = pitch_lag_int;
335
25024
    pitch_lag_int += pitch_lag_frac > 0;
336
337
    /* Calculate the pitch vector by interpolating the past excitation at the
338
       pitch lag using a hamming windowed sinc function */
339
50048
    ctx->acelpf_ctx.acelp_interpolatef(exc,
340
25024
                          exc + 1 - pitch_lag_int,
341
                          ac_inter, 4,
342
25024
                          pitch_lag_frac + (pitch_lag_frac > 0 ? 0 : 4),
343
                          LP_ORDER, AMRWB_SFR_SIZE + 1);
344
345
    /* Check which pitch signal path should be used
346
     * 6k60 and 8k85 modes have the ltp flag set to 0 */
347
25024
    if (amr_subframe->ltp) {
348
7110
        memcpy(ctx->pitch_vector, exc, AMRWB_SFR_SIZE * sizeof(float));
349
    } else {
350
1164410
        for (i = 0; i < AMRWB_SFR_SIZE; i++)
351
1146496
            ctx->pitch_vector[i] = 0.18 * exc[i - 1] + 0.64 * exc[i] +
352
1146496
                                   0.18 * exc[i + 1];
353
17914
        memcpy(exc, ctx->pitch_vector, AMRWB_SFR_SIZE * sizeof(float));
354
    }
355
25024
}
356
357
/** Get x bits in the index interval [lsb,lsb+len-1] inclusive */
358
#define BIT_STR(x,lsb,len) av_mod_uintp2((x) >> (lsb), (len))
359
360
/** Get the bit at specified position */
361
#define BIT_POS(x, p) (((x) >> (p)) & 1)
362
363
/**
364
 * The next six functions decode_[i]p_track decode exactly i pulses
365
 * positions and amplitudes (-1 or 1) in a subframe track using
366
 * an encoded pulse indexing (TS 26.190 section 5.8.2).
367
 *
368
 * The results are given in out[], in which a negative number means
369
 * amplitude -1 and vice versa (i.e., ampl(x) = x / abs(x) ).
370
 *
371
 * @param[out] out                 Output buffer (writes i elements)
372
 * @param[in]  code                Pulse index (no. of bits varies, see below)
373
 * @param[in]  m                   (log2) Number of potential positions
374
 * @param[in]  off                 Offset for decoded positions
375
 */
376
105998
static inline void decode_1p_track(int *out, int code, int m, int off)
377
{
378
105998
    int pos = BIT_STR(code, 0, m) + off; ///code: m+1 bits
379
380
105998
    out[0] = BIT_POS(code, m) ? -pos : pos;
381
105998
}
382
383
146937
static inline void decode_2p_track(int *out, int code, int m, int off) ///code: 2m+1 bits
384
{
385
146937
    int pos0 = BIT_STR(code, m, m) + off;
386
146937
    int pos1 = BIT_STR(code, 0, m) + off;
387
388
146937
    out[0] = BIT_POS(code, 2*m) ? -pos0 : pos0;
389
146937
    out[1] = BIT_POS(code, 2*m) ? -pos1 : pos1;
390
146937
    out[1] = pos0 > pos1 ? -out[1] : out[1];
391
146937
}
392
393
68636
static void decode_3p_track(int *out, int code, int m, int off) ///code: 3m+1 bits
394
{
395
68636
    int half_2p = BIT_POS(code, 2*m - 1) << (m - 1);
396
397
68636
    decode_2p_track(out, BIT_STR(code, 0, 2*m - 1),
398
                    m - 1, off + half_2p);
399
68636
    decode_1p_track(out + 2, BIT_STR(code, 2*m, m + 1), m, off);
400
68636
}
401
402
32497
static void decode_4p_track(int *out, int code, int m, int off) ///code: 4m bits
403
{
404
    int half_4p, subhalf_2p;
405
32497
    int b_offset = 1 << (m - 1);
406
407

32497
    switch (BIT_STR(code, 4*m - 2, 2)) { /* case ID (2 bits) */
408
3597
    case 0: /* 0 pulses in A, 4 pulses in B or vice versa */
409
3597
        half_4p = BIT_POS(code, 4*m - 3) << (m - 1); // which has 4 pulses
410
3597
        subhalf_2p = BIT_POS(code, 2*m - 3) << (m - 2);
411
412
3597
        decode_2p_track(out, BIT_STR(code, 0, 2*m - 3),
413
3597
                        m - 2, off + half_4p + subhalf_2p);
414
3597
        decode_2p_track(out + 2, BIT_STR(code, 2*m - 2, 2*m - 1),
415
                        m - 1, off + half_4p);
416
3597
        break;
417
7976
    case 1: /* 1 pulse in A, 3 pulses in B */
418
7976
        decode_1p_track(out, BIT_STR(code,  3*m - 2, m),
419
                        m - 1, off);
420
7976
        decode_3p_track(out + 1, BIT_STR(code, 0, 3*m - 2),
421
                        m - 1, off + b_offset);
422
7976
        break;
423
12780
    case 2: /* 2 pulses in each half */
424
12780
        decode_2p_track(out, BIT_STR(code, 2*m - 1, 2*m - 1),
425
                        m - 1, off);
426
12780
        decode_2p_track(out + 2, BIT_STR(code, 0, 2*m - 1),
427
                        m - 1, off + b_offset);
428
12780
        break;
429
8144
    case 3: /* 3 pulses in A, 1 pulse in B */
430
8144
        decode_3p_track(out, BIT_STR(code, m, 3*m - 2),
431
                        m - 1, off);
432
8144
        decode_1p_track(out + 3, BIT_STR(code, 0, m),
433
                        m - 1, off + b_offset);
434
8144
        break;
435
    }
436
32497
}
437
438
13050
static void decode_5p_track(int *out, int code, int m, int off) ///code: 5m bits
439
{
440
13050
    int half_3p = BIT_POS(code, 5*m - 1) << (m - 1);
441
442
13050
    decode_3p_track(out, BIT_STR(code, 2*m + 1, 3*m - 2),
443
                    m - 1, off + half_3p);
444
445
13050
    decode_2p_track(out + 3, BIT_STR(code, 0, 2*m + 1), m, off);
446
13050
}
447
448
42752
static void decode_6p_track(int *out, int code, int m, int off) ///code: 6m-2 bits
449
{
450
42752
    int b_offset = 1 << (m - 1);
451
    /* which half has more pulses in cases 0 to 2 */
452
42752
    int half_more  = BIT_POS(code, 6*m - 5) << (m - 1);
453
42752
    int half_other = b_offset - half_more;
454
455

42752
    switch (BIT_STR(code, 6*m - 4, 2)) { /* case ID (2 bits) */
456
1288
    case 0: /* 0 pulses in A, 6 pulses in B or vice versa */
457
1288
        decode_1p_track(out, BIT_STR(code, 0, m),
458
                        m - 1, off + half_more);
459
1288
        decode_5p_track(out + 1, BIT_STR(code, m, 5*m - 5),
460
                        m - 1, off + half_more);
461
1288
        break;
462
7666
    case 1: /* 1 pulse in A, 5 pulses in B or vice versa */
463
7666
        decode_1p_track(out, BIT_STR(code, 0, m),
464
                        m - 1, off + half_other);
465
7666
        decode_5p_track(out + 1, BIT_STR(code, m, 5*m - 5),
466
                        m - 1, off + half_more);
467
7666
        break;
468
20209
    case 2: /* 2 pulses in A, 4 pulses in B or vice versa */
469
20209
        decode_2p_track(out, BIT_STR(code, 0, 2*m - 1),
470
                        m - 1, off + half_other);
471
20209
        decode_4p_track(out + 2, BIT_STR(code, 2*m - 1, 4*m - 4),
472
                        m - 1, off + half_more);
473
20209
        break;
474
13589
    case 3: /* 3 pulses in A, 3 pulses in B */
475
13589
        decode_3p_track(out, BIT_STR(code, 3*m - 2, 3*m - 2),
476
                        m - 1, off);
477
13589
        decode_3p_track(out + 3, BIT_STR(code, 0, 3*m - 2),
478
                        m - 1, off + b_offset);
479
13589
        break;
480
    }
481
42752
}
482
483
/**
484
 * Decode the algebraic codebook index to pulse positions and signs,
485
 * then construct the algebraic codebook vector.
486
 *
487
 * @param[out] fixed_vector        Buffer for the fixed codebook excitation
488
 * @param[in]  pulse_hi            MSBs part of the pulse index array (higher modes only)
489
 * @param[in]  pulse_lo            LSBs part of the pulse index array
490
 * @param[in]  mode                Mode of the current frame
491
 */
492
25024
static void decode_fixed_vector(float *fixed_vector, const uint16_t *pulse_hi,
493
                                const uint16_t *pulse_lo, const enum Mode mode)
494
{
495
    /* sig_pos stores for each track the decoded pulse position indexes
496
     * (1-based) multiplied by its corresponding amplitude (+1 or -1) */
497
    int sig_pos[4][6];
498
25024
    int spacing = (mode == MODE_6k60) ? 2 : 4;
499
    int i, j;
500
501


25024
    switch (mode) {
502
2048
    case MODE_6k60:
503
6144
        for (i = 0; i < 2; i++)
504
4096
            decode_1p_track(sig_pos[i], pulse_lo[i], 5, 1);
505
2048
        break;
506
2048
    case MODE_8k85:
507
10240
        for (i = 0; i < 4; i++)
508
8192
            decode_1p_track(sig_pos[i], pulse_lo[i], 4, 1);
509
2048
        break;
510
2048
    case MODE_12k65:
511
10240
        for (i = 0; i < 4; i++)
512
8192
            decode_2p_track(sig_pos[i], pulse_lo[i], 4, 1);
513
2048
        break;
514
2048
    case MODE_14k25:
515
6144
        for (i = 0; i < 2; i++)
516
4096
            decode_3p_track(sig_pos[i], pulse_lo[i], 4, 1);
517
6144
        for (i = 2; i < 4; i++)
518
4096
            decode_2p_track(sig_pos[i], pulse_lo[i], 4, 1);
519
2048
        break;
520
2048
    case MODE_15k85:
521
10240
        for (i = 0; i < 4; i++)
522
8192
            decode_3p_track(sig_pos[i], pulse_lo[i], 4, 1);
523
2048
        break;
524
2048
    case MODE_18k25:
525
10240
        for (i = 0; i < 4; i++)
526
8192
            decode_4p_track(sig_pos[i], (int) pulse_lo[i] +
527
8192
                           ((int) pulse_hi[i] << 14), 4, 1);
528
2048
        break;
529
2048
    case MODE_19k85:
530
6144
        for (i = 0; i < 2; i++)
531
4096
            decode_5p_track(sig_pos[i], (int) pulse_lo[i] +
532
4096
                           ((int) pulse_hi[i] << 10), 4, 1);
533
6144
        for (i = 2; i < 4; i++)
534
4096
            decode_4p_track(sig_pos[i], (int) pulse_lo[i] +
535
4096
                           ((int) pulse_hi[i] << 14), 4, 1);
536
2048
        break;
537
10688
    case MODE_23k05:
538
    case MODE_23k85:
539
53440
        for (i = 0; i < 4; i++)
540
42752
            decode_6p_track(sig_pos[i], (int) pulse_lo[i] +
541
42752
                           ((int) pulse_hi[i] << 11), 4, 1);
542
10688
        break;
543
    }
544
545
25024
    memset(fixed_vector, 0, sizeof(float) * AMRWB_SFR_SIZE);
546
547
125120
    for (i = 0; i < 4; i++)
548
499968
        for (j = 0; j < pulses_nb_per_mode_tr[mode][i]; j++) {
549
399872
            int pos = (FFABS(sig_pos[i][j]) - 1) * spacing + i;
550
551
399872
            fixed_vector[pos] += sig_pos[i][j] < 0 ? -1.0 : 1.0;
552
        }
553
25024
}
554
555
/**
556
 * Decode pitch gain and fixed gain correction factor.
557
 *
558
 * @param[in]  vq_gain             Vector-quantized index for gains
559
 * @param[in]  mode                Mode of the current frame
560
 * @param[out] fixed_gain_factor   Decoded fixed gain correction factor
561
 * @param[out] pitch_gain          Decoded pitch gain
562
 */
563
25024
static void decode_gains(const uint8_t vq_gain, const enum Mode mode,
564
                         float *fixed_gain_factor, float *pitch_gain)
565
{
566
25024
    const int16_t *gains = (mode <= MODE_8k85 ? qua_gain_6b[vq_gain] :
567
20928
                                                qua_gain_7b[vq_gain]);
568
569
25024
    *pitch_gain        = gains[0] * (1.0f / (1 << 14));
570
25024
    *fixed_gain_factor = gains[1] * (1.0f / (1 << 11));
571
25024
}
572
573
/**
574
 * Apply pitch sharpening filters to the fixed codebook vector.
575
 *
576
 * @param[in]     ctx              The context
577
 * @param[in,out] fixed_vector     Fixed codebook excitation
578
 */
579
// XXX: Spec states this procedure should be applied when the pitch
580
// lag is less than 64, but this checking seems absent in reference and AMR-NB
581
25024
static void pitch_sharpening(AMRWBContext *ctx, float *fixed_vector)
582
{
583
    int i;
584
585
    /* Tilt part */
586
1601536
    for (i = AMRWB_SFR_SIZE - 1; i != 0; i--)
587
1576512
        fixed_vector[i] -= fixed_vector[i - 1] * ctx->tilt_coef;
588
589
    /* Periodicity enhancement part */
590
185744
    for (i = ctx->pitch_lag_int; i < AMRWB_SFR_SIZE; i++)
591
160720
        fixed_vector[i] += fixed_vector[i - ctx->pitch_lag_int] * 0.85;
592
25024
}
593
594
/**
595
 * Calculate the voicing factor (-1.0 = unvoiced to 1.0 = voiced).
596
 *
597
 * @param[in] p_vector, f_vector   Pitch and fixed excitation vectors
598
 * @param[in] p_gain, f_gain       Pitch and fixed gains
599
 * @param[in] ctx                  The context
600
 */
601
// XXX: There is something wrong with the precision here! The magnitudes
602
// of the energies are not correct. Please check the reference code carefully
603
25024
static float voice_factor(float *p_vector, float p_gain,
604
                          float *f_vector, float f_gain,
605
                          CELPMContext *ctx)
606
{
607
25024
    double p_ener = (double) ctx->dot_productf(p_vector, p_vector,
608
25024
                                                          AMRWB_SFR_SIZE) *
609
25024
                    p_gain * p_gain;
610
25024
    double f_ener = (double) ctx->dot_productf(f_vector, f_vector,
611
25024
                                                          AMRWB_SFR_SIZE) *
612
25024
                    f_gain * f_gain;
613
614
25024
    return (p_ener - f_ener) / (p_ener + f_ener + 0.01);
615
}
616
617
/**
618
 * Reduce fixed vector sparseness by smoothing with one of three IR filters,
619
 * also known as "adaptive phase dispersion".
620
 *
621
 * @param[in]     ctx              The context
622
 * @param[in,out] fixed_vector     Unfiltered fixed vector
623
 * @param[out]    buf              Space for modified vector if necessary
624
 *
625
 * @return The potentially overwritten filtered fixed vector address
626
 */
627
25024
static float *anti_sparseness(AMRWBContext *ctx,
628
                              float *fixed_vector, float *buf)
629
{
630
    int ir_filter_nr;
631
632
25024
    if (ctx->fr_cur_mode > MODE_8k85) // no filtering in higher modes
633
20928
        return fixed_vector;
634
635
4096
    if (ctx->pitch_gain[0] < 0.6) {
636
2254
        ir_filter_nr = 0;      // strong filtering
637
1842
    } else if (ctx->pitch_gain[0] < 0.9) {
638
712
        ir_filter_nr = 1;      // medium filtering
639
    } else
640
1130
        ir_filter_nr = 2;      // no filtering
641
642
    /* detect 'onset' */
643
4096
    if (ctx->fixed_gain[0] > 3.0 * ctx->fixed_gain[1]) {
644
57
        if (ir_filter_nr < 2)
645
35
            ir_filter_nr++;
646
    } else {
647
4039
        int i, count = 0;
648
649
28273
        for (i = 0; i < 6; i++)
650
24234
            if (ctx->pitch_gain[i] < 0.6)
651
13350
                count++;
652
653
4039
        if (count > 2)
654
2720
            ir_filter_nr = 0;
655
656
4039
        if (ir_filter_nr > ctx->prev_ir_filter_nr + 1)
657
83
            ir_filter_nr--;
658
    }
659
660
    /* update ir filter strength history */
661
4096
    ctx->prev_ir_filter_nr = ir_filter_nr;
662
663
4096
    ir_filter_nr += (ctx->fr_cur_mode == MODE_8k85);
664
665
4096
    if (ir_filter_nr < 2) {
666
        int i;
667
3191
        const float *coef = ir_filters_lookup[ir_filter_nr];
668
669
        /* Circular convolution code in the reference
670
         * decoder was modified to avoid using one
671
         * extra array. The filtered vector is given by:
672
         *
673
         * c2(n) = sum(i,0,len-1){ c(i) * coef( (n - i + len) % len ) }
674
         */
675
676
3191
        memset(buf, 0, sizeof(float) * AMRWB_SFR_SIZE);
677
207415
        for (i = 0; i < AMRWB_SFR_SIZE; i++)
678
204224
            if (fixed_vector[i])
679
21429
                ff_celp_circ_addf(buf, buf, coef, i, fixed_vector[i],
680
                                  AMRWB_SFR_SIZE);
681
3191
        fixed_vector = buf;
682
    }
683
684
4096
    return fixed_vector;
685
}
686
687
/**
688
 * Calculate a stability factor {teta} based on distance between
689
 * current and past isf. A value of 1 shows maximum signal stability.
690
 */
691
6256
static float stability_factor(const float *isf, const float *isf_past)
692
{
693
    int i;
694
6256
    float acc = 0.0;
695
696
100096
    for (i = 0; i < LP_ORDER - 1; i++)
697
93840
        acc += (isf[i] - isf_past[i]) * (isf[i] - isf_past[i]);
698
699
    // XXX: This part is not so clear from the reference code
700
    // the result is more accurate changing the "/ 256" to "* 512"
701
6256
    return FFMAX(0.0, 1.25 - acc * 0.8 * 512);
702
}
703
704
/**
705
 * Apply a non-linear fixed gain smoothing in order to reduce
706
 * fluctuation in the energy of excitation.
707
 *
708
 * @param[in]     fixed_gain       Unsmoothed fixed gain
709
 * @param[in,out] prev_tr_gain     Previous threshold gain (updated)
710
 * @param[in]     voice_fac        Frame voicing factor
711
 * @param[in]     stab_fac         Frame stability factor
712
 *
713
 * @return The smoothed gain
714
 */
715
25024
static float noise_enhancer(float fixed_gain, float *prev_tr_gain,
716
                            float voice_fac,  float stab_fac)
717
{
718
25024
    float sm_fac = 0.5 * (1 - voice_fac) * stab_fac;
719
    float g0;
720
721
    // XXX: the following fixed-point constants used to in(de)crement
722
    // gain by 1.5dB were taken from the reference code, maybe it could
723
    // be simpler
724
25024
    if (fixed_gain < *prev_tr_gain) {
725
13114
        g0 = FFMIN(*prev_tr_gain, fixed_gain + fixed_gain *
726
                     (6226 * (1.0f / (1 << 15)))); // +1.5 dB
727
    } else
728
11910
        g0 = FFMAX(*prev_tr_gain, fixed_gain *
729
                    (27536 * (1.0f / (1 << 15)))); // -1.5 dB
730
731
25024
    *prev_tr_gain = g0; // update next frame threshold
732
733
25024
    return sm_fac * g0 + (1 - sm_fac) * fixed_gain;
734
}
735
736
/**
737
 * Filter the fixed_vector to emphasize the higher frequencies.
738
 *
739
 * @param[in,out] fixed_vector     Fixed codebook vector
740
 * @param[in]     voice_fac        Frame voicing factor
741
 */
742
25024
static void pitch_enhancer(float *fixed_vector, float voice_fac)
743
{
744
    int i;
745
25024
    float cpe  = 0.125 * (1 + voice_fac);
746
25024
    float last = fixed_vector[0]; // holds c(i - 1)
747
748
25024
    fixed_vector[0] -= cpe * fixed_vector[1];
749
750
1576512
    for (i = 1; i < AMRWB_SFR_SIZE - 1; i++) {
751
1551488
        float cur = fixed_vector[i];
752
753
1551488
        fixed_vector[i] -= cpe * (last + fixed_vector[i + 1]);
754
1551488
        last = cur;
755
    }
756
757
25024
    fixed_vector[AMRWB_SFR_SIZE - 1] -= cpe * last;
758
25024
}
759
760
/**
761
 * Conduct 16th order linear predictive coding synthesis from excitation.
762
 *
763
 * @param[in]     ctx              Pointer to the AMRWBContext
764
 * @param[in]     lpc              Pointer to the LPC coefficients
765
 * @param[out]    excitation       Buffer for synthesis final excitation
766
 * @param[in]     fixed_gain       Fixed codebook gain for synthesis
767
 * @param[in]     fixed_vector     Algebraic codebook vector
768
 * @param[in,out] samples          Pointer to the output samples and memory
769
 */
770
25024
static void synthesis(AMRWBContext *ctx, float *lpc, float *excitation,
771
                      float fixed_gain, const float *fixed_vector,
772
                      float *samples)
773
{
774
25024
    ctx->acelpv_ctx.weighted_vector_sumf(excitation, ctx->pitch_vector, fixed_vector,
775
                            ctx->pitch_gain[0], fixed_gain, AMRWB_SFR_SIZE);
776
777
    /* emphasize pitch vector contribution in low bitrate modes */
778

25024
    if (ctx->pitch_gain[0] > 0.5 && ctx->fr_cur_mode <= MODE_8k85) {
779
        int i;
780
1895
        float energy = ctx->celpm_ctx.dot_productf(excitation, excitation,
781
                                                    AMRWB_SFR_SIZE);
782
783
        // XXX: Weird part in both ref code and spec. A unknown parameter
784
        // {beta} seems to be identical to the current pitch gain
785
1895
        float pitch_factor = 0.25 * ctx->pitch_gain[0] * ctx->pitch_gain[0];
786
787
123175
        for (i = 0; i < AMRWB_SFR_SIZE; i++)
788
121280
            excitation[i] += pitch_factor * ctx->pitch_vector[i];
789
790
1895
        ff_scale_vector_to_given_sum_of_squares(excitation, excitation,
791
                                                energy, AMRWB_SFR_SIZE);
792
    }
793
794
25024
    ctx->celpf_ctx.celp_lp_synthesis_filterf(samples, lpc, excitation,
795
                                 AMRWB_SFR_SIZE, LP_ORDER);
796
25024
}
797
798
/**
799
 * Apply to synthesis a de-emphasis filter of the form:
800
 * H(z) = 1 / (1 - m * z^-1)
801
 *
802
 * @param[out]    out              Output buffer
803
 * @param[in]     in               Input samples array with in[-1]
804
 * @param[in]     m                Filter coefficient
805
 * @param[in,out] mem              State from last filtering
806
 */
807
25024
static void de_emphasis(float *out, float *in, float m, float mem[1])
808
{
809
    int i;
810
811
25024
    out[0] = in[0] + m * mem[0];
812
813
1601536
    for (i = 1; i < AMRWB_SFR_SIZE; i++)
814
1576512
         out[i] = in[i] + out[i - 1] * m;
815
816
25024
    mem[0] = out[AMRWB_SFR_SIZE - 1];
817
25024
}
818
819
/**
820
 * Upsample a signal by 5/4 ratio (from 12.8kHz to 16kHz) using
821
 * a FIR interpolation filter. Uses past data from before *in address.
822
 *
823
 * @param[out] out                 Buffer for interpolated signal
824
 * @param[in]  in                  Current signal data (length 0.8*o_size)
825
 * @param[in]  o_size              Output signal length
826
 * @param[in] ctx                  The context
827
 */
828
25024
static void upsample_5_4(float *out, const float *in, int o_size, CELPMContext *ctx)
829
{
830
25024
    const float *in0 = in - UPS_FIR_SIZE + 1;
831
    int i, j, k;
832
25024
    int int_part = 0, frac_part;
833
834
25024
    i = 0;
835
425408
    for (j = 0; j < o_size / 5; j++) {
836
400384
        out[i] = in[int_part];
837
400384
        frac_part = 4;
838
400384
        i++;
839
840
2001920
        for (k = 1; k < 5; k++) {
841
3203072
            out[i] = ctx->dot_productf(in0 + int_part,
842
1601536
                                                  upsample_fir[4 - frac_part],
843
                                                  UPS_MEM_SIZE);
844
1601536
            int_part++;
845
1601536
            frac_part--;
846
1601536
            i++;
847
        }
848
    }
849
25024
}
850
851
/**
852
 * Calculate the high-band gain based on encoded index (23k85 mode) or
853
 * on the low-band speech signal and the Voice Activity Detection flag.
854
 *
855
 * @param[in] ctx                  The context
856
 * @param[in] synth                LB speech synthesis at 12.8k
857
 * @param[in] hb_idx               Gain index for mode 23k85 only
858
 * @param[in] vad                  VAD flag for the frame
859
 */
860
25024
static float find_hb_gain(AMRWBContext *ctx, const float *synth,
861
                          uint16_t hb_idx, uint8_t vad)
862
{
863
25024
    int wsp = (vad > 0);
864
    float tilt;
865
    float tmp;
866
867
25024
    if (ctx->fr_cur_mode == MODE_23k85)
868
8640
        return qua_hb_gain[hb_idx] * (1.0f / (1 << 14));
869
870
16384
    tmp = ctx->celpm_ctx.dot_productf(synth, synth + 1, AMRWB_SFR_SIZE - 1);
871
872
16384
    if (tmp > 0) {
873
15621
        tilt = tmp / ctx->celpm_ctx.dot_productf(synth, synth, AMRWB_SFR_SIZE);
874
    } else
875
763
        tilt = 0;
876
877
    /* return gain bounded by [0.1, 1.0] */
878
16384
    return av_clipf((1.0 - tilt) * (1.25 - 0.25 * wsp), 0.1, 1.0);
879
}
880
881
/**
882
 * Generate the high-band excitation with the same energy from the lower
883
 * one and scaled by the given gain.
884
 *
885
 * @param[in]  ctx                 The context
886
 * @param[out] hb_exc              Buffer for the excitation
887
 * @param[in]  synth_exc           Low-band excitation used for synthesis
888
 * @param[in]  hb_gain             Wanted excitation gain
889
 */
890
25024
static void scaled_hb_excitation(AMRWBContext *ctx, float *hb_exc,
891
                                 const float *synth_exc, float hb_gain)
892
{
893
    int i;
894
25024
    float energy = ctx->celpm_ctx.dot_productf(synth_exc, synth_exc,
895
                                                AMRWB_SFR_SIZE);
896
897
    /* Generate a white-noise excitation */
898
2026944
    for (i = 0; i < AMRWB_SFR_SIZE_16k; i++)
899
2001920
        hb_exc[i] = 32768.0 - (uint16_t) av_lfg_get(&ctx->prng);
900
901
25024
    ff_scale_vector_to_given_sum_of_squares(hb_exc, hb_exc,
902
25024
                                            energy * hb_gain * hb_gain,
903
                                            AMRWB_SFR_SIZE_16k);
904
25024
}
905
906
/**
907
 * Calculate the auto-correlation for the ISF difference vector.
908
 */
909
6144
static float auto_correlation(float *diff_isf, float mean, int lag)
910
{
911
    int i;
912
6144
    float sum = 0.0;
913
914
49152
    for (i = 7; i < LP_ORDER - 2; i++) {
915
43008
        float prod = (diff_isf[i] - mean) * (diff_isf[i - lag] - mean);
916
43008
        sum += prod * prod;
917
    }
918
6144
    return sum;
919
}
920
921
/**
922
 * Extrapolate a ISF vector to the 16kHz range (20th order LP)
923
 * used at mode 6k60 LP filter for the high frequency band.
924
 *
925
 * @param[out] isf Buffer for extrapolated isf; contains LP_ORDER
926
 *                 values on input
927
 */
928
2048
static void extrapolate_isf(float isf[LP_ORDER_16k])
929
{
930
    float diff_isf[LP_ORDER - 2], diff_mean;
931
    float corr_lag[3];
932
    float est, scale;
933
    int i, j, i_max_corr;
934
935
2048
    isf[LP_ORDER_16k - 1] = isf[LP_ORDER - 1];
936
937
    /* Calculate the difference vector */
938
30720
    for (i = 0; i < LP_ORDER - 2; i++)
939
28672
        diff_isf[i] = isf[i + 1] - isf[i];
940
941
2048
    diff_mean = 0.0;
942
26624
    for (i = 2; i < LP_ORDER - 2; i++)
943
24576
        diff_mean += diff_isf[i] * (1.0f / (LP_ORDER - 4));
944
945
    /* Find which is the maximum autocorrelation */
946
2048
    i_max_corr = 0;
947
8192
    for (i = 0; i < 3; i++) {
948
6144
        corr_lag[i] = auto_correlation(diff_isf, diff_mean, i + 2);
949
950
6144
        if (corr_lag[i] > corr_lag[i_max_corr])
951
2154
            i_max_corr = i;
952
    }
953
2048
    i_max_corr++;
954
955
10240
    for (i = LP_ORDER - 1; i < LP_ORDER_16k - 1; i++)
956
8192
        isf[i] = isf[i - 1] + isf[i - 1 - i_max_corr]
957
8192
                            - isf[i - 2 - i_max_corr];
958
959
    /* Calculate an estimate for ISF(18) and scale ISF based on the error */
960
2048
    est   = 7965 + (isf[2] - isf[3] - isf[4]) / 6.0;
961
2048
    scale = 0.5 * (FFMIN(est, 7600) - isf[LP_ORDER - 2]) /
962
2048
            (isf[LP_ORDER_16k - 2] - isf[LP_ORDER - 2]);
963
964
10240
    for (i = LP_ORDER - 1, j = 0; i < LP_ORDER_16k - 1; i++, j++)
965
8192
        diff_isf[j] = scale * (isf[i] - isf[i - 1]);
966
967
    /* Stability insurance */
968
8192
    for (i = 1; i < LP_ORDER_16k - LP_ORDER; i++)
969
6144
        if (diff_isf[i] + diff_isf[i - 1] < 5.0) {
970
            if (diff_isf[i] > diff_isf[i - 1]) {
971
                diff_isf[i - 1] = 5.0 - diff_isf[i];
972
            } else
973
                diff_isf[i] = 5.0 - diff_isf[i - 1];
974
        }
975
976
10240
    for (i = LP_ORDER - 1, j = 0; i < LP_ORDER_16k - 1; i++, j++)
977
8192
        isf[i] = isf[i - 1] + diff_isf[j] * (1.0f / (1 << 15));
978
979
    /* Scale the ISF vector for 16000 Hz */
980
40960
    for (i = 0; i < LP_ORDER_16k - 1; i++)
981
38912
        isf[i] *= 0.8;
982
2048
}
983
984
/**
985
 * Spectral expand the LP coefficients using the equation:
986
 *   y[i] = x[i] * (gamma ** i)
987
 *
988
 * @param[out] out                 Output buffer (may use input array)
989
 * @param[in]  lpc                 LP coefficients array
990
 * @param[in]  gamma               Weighting factor
991
 * @param[in]  size                LP array size
992
 */
993
25024
static void lpc_weighting(float *out, const float *lpc, float gamma, int size)
994
{
995
    int i;
996
25024
    float fac = gamma;
997
998
433600
    for (i = 0; i < size; i++) {
999
408576
        out[i] = lpc[i] * fac;
1000
408576
        fac   *= gamma;
1001
    }
1002
25024
}
1003
1004
/**
1005
 * Conduct 20th order linear predictive coding synthesis for the high
1006
 * frequency band excitation at 16kHz.
1007
 *
1008
 * @param[in]     ctx              The context
1009
 * @param[in]     subframe         Current subframe index (0 to 3)
1010
 * @param[in,out] samples          Pointer to the output speech samples
1011
 * @param[in]     exc              Generated white-noise scaled excitation
1012
 * @param[in]     isf              Current frame isf vector
1013
 * @param[in]     isf_past         Past frame final isf vector
1014
 */
1015
25024
static void hb_synthesis(AMRWBContext *ctx, int subframe, float *samples,
1016
                         const float *exc, const float *isf, const float *isf_past)
1017
{
1018
    float hb_lpc[LP_ORDER_16k];
1019
25024
    enum Mode mode = ctx->fr_cur_mode;
1020
1021
25024
    if (mode == MODE_6k60) {
1022
        float e_isf[LP_ORDER_16k]; // ISF vector for extrapolation
1023
        double e_isp[LP_ORDER_16k];
1024
1025
2048
        ctx->acelpv_ctx.weighted_vector_sumf(e_isf, isf_past, isf, isfp_inter[subframe],
1026
2048
                                1.0 - isfp_inter[subframe], LP_ORDER);
1027
1028
2048
        extrapolate_isf(e_isf);
1029
1030
2048
        e_isf[LP_ORDER_16k - 1] *= 2.0;
1031
2048
        ff_acelp_lsf2lspd(e_isp, e_isf, LP_ORDER_16k);
1032
2048
        ff_amrwb_lsp2lpc(e_isp, hb_lpc, LP_ORDER_16k);
1033
1034
2048
        lpc_weighting(hb_lpc, hb_lpc, 0.9, LP_ORDER_16k);
1035
    } else {
1036
22976
        lpc_weighting(hb_lpc, ctx->lp_coef[subframe], 0.6, LP_ORDER);
1037
    }
1038
1039
25024
    ctx->celpf_ctx.celp_lp_synthesis_filterf(samples, hb_lpc, exc, AMRWB_SFR_SIZE_16k,
1040
                                 (mode == MODE_6k60) ? LP_ORDER_16k : LP_ORDER);
1041
25024
}
1042
1043
/**
1044
 * Apply a 15th order filter to high-band samples.
1045
 * The filter characteristic depends on the given coefficients.
1046
 *
1047
 * @param[out]    out              Buffer for filtered output
1048
 * @param[in]     fir_coef         Filter coefficients
1049
 * @param[in,out] mem              State from last filtering (updated)
1050
 * @param[in]     in               Input speech data (high-band)
1051
 *
1052
 * @remark It is safe to pass the same array in in and out parameters
1053
 */
1054
1055
#ifndef hb_fir_filter
1056
33664
static void hb_fir_filter(float *out, const float fir_coef[HB_FIR_SIZE + 1],
1057
                          float mem[HB_FIR_SIZE], const float *in)
1058
{
1059
    int i, j;
1060
    float data[AMRWB_SFR_SIZE_16k + HB_FIR_SIZE]; // past and current samples
1061
1062
33664
    memcpy(data, mem, HB_FIR_SIZE * sizeof(float));
1063
33664
    memcpy(data + HB_FIR_SIZE, in, AMRWB_SFR_SIZE_16k * sizeof(float));
1064
1065
2726784
    for (i = 0; i < AMRWB_SFR_SIZE_16k; i++) {
1066
2693120
        out[i] = 0.0;
1067
86179840
        for (j = 0; j <= HB_FIR_SIZE; j++)
1068
83486720
            out[i] += data[i + j] * fir_coef[j];
1069
    }
1070
1071
33664
    memcpy(mem, data + AMRWB_SFR_SIZE_16k, HB_FIR_SIZE * sizeof(float));
1072
33664
}
1073
#endif /* hb_fir_filter */
1074
1075
/**
1076
 * Update context state before the next subframe.
1077
 */
1078
25024
static void update_sub_state(AMRWBContext *ctx)
1079
{
1080
25024
    memmove(&ctx->excitation_buf[0], &ctx->excitation_buf[AMRWB_SFR_SIZE],
1081
            (AMRWB_P_DELAY_MAX + LP_ORDER + 1) * sizeof(float));
1082
1083
25024
    memmove(&ctx->pitch_gain[1], &ctx->pitch_gain[0], 5 * sizeof(float));
1084
25024
    memmove(&ctx->fixed_gain[1], &ctx->fixed_gain[0], 1 * sizeof(float));
1085
1086
25024
    memmove(&ctx->samples_az[0], &ctx->samples_az[AMRWB_SFR_SIZE],
1087
            LP_ORDER * sizeof(float));
1088
25024
    memmove(&ctx->samples_up[0], &ctx->samples_up[AMRWB_SFR_SIZE],
1089
            UPS_MEM_SIZE * sizeof(float));
1090
25024
    memmove(&ctx->samples_hb[0], &ctx->samples_hb[AMRWB_SFR_SIZE_16k],
1091
            LP_ORDER_16k * sizeof(float));
1092
25024
}
1093
1094
6256
static int amrwb_decode_frame(AVCodecContext *avctx, void *data,
1095
                              int *got_frame_ptr, AVPacket *avpkt)
1096
{
1097
6256
    AMRWBContext *ctx  = avctx->priv_data;
1098
6256
    AVFrame *frame     = data;
1099
6256
    AMRWBFrame   *cf   = &ctx->frame;
1100
6256
    const uint8_t *buf = avpkt->data;
1101
6256
    int buf_size       = avpkt->size;
1102
    int expected_fr_size, header_size;
1103
    float *buf_out;
1104
    float spare_vector[AMRWB_SFR_SIZE];      // extra stack space to hold result from anti-sparseness processing
1105
    float fixed_gain_factor;                 // fixed gain correction factor (gamma)
1106
    float *synth_fixed_vector;               // pointer to the fixed vector that synthesis should use
1107
    float synth_fixed_gain;                  // the fixed gain that synthesis should use
1108
    float voice_fac, stab_fac;               // parameters used for gain smoothing
1109
    float synth_exc[AMRWB_SFR_SIZE];         // post-processed excitation for synthesis
1110
    float hb_exc[AMRWB_SFR_SIZE_16k];        // excitation for the high frequency band
1111
    float hb_samples[AMRWB_SFR_SIZE_16k];    // filtered high-band samples from synthesis
1112
    float hb_gain;
1113
    int sub, i, ret;
1114
1115
    /* get output buffer */
1116
6256
    frame->nb_samples = 4 * AMRWB_SFR_SIZE_16k;
1117
6256
    if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
1118
        return ret;
1119
6256
    buf_out = (float *)frame->data[0];
1120
1121
6256
    header_size      = decode_mime_header(ctx, buf);
1122
6256
    if (ctx->fr_cur_mode > MODE_SID) {
1123
        av_log(avctx, AV_LOG_ERROR,
1124
               "Invalid mode %d\n", ctx->fr_cur_mode);
1125
        return AVERROR_INVALIDDATA;
1126
    }
1127
6256
    expected_fr_size = ((cf_sizes_wb[ctx->fr_cur_mode] + 7) >> 3) + 1;
1128
1129
6256
    if (buf_size < expected_fr_size) {
1130
        av_log(avctx, AV_LOG_ERROR,
1131
            "Frame too small (%d bytes). Truncated file?\n", buf_size);
1132
        *got_frame_ptr = 0;
1133
        return AVERROR_INVALIDDATA;
1134
    }
1135
1136

6256
    if (!ctx->fr_quality || ctx->fr_cur_mode > MODE_SID)
1137
        av_log(avctx, AV_LOG_ERROR, "Encountered a bad or corrupted frame\n");
1138
1139
6256
    if (ctx->fr_cur_mode == MODE_SID) { /* Comfort noise frame */
1140
        avpriv_request_sample(avctx, "SID mode");
1141
        return AVERROR_PATCHWELCOME;
1142
    }
1143
1144
6256
    ff_amr_bit_reorder((uint16_t *) &ctx->frame, sizeof(AMRWBFrame),
1145
6256
        buf + header_size, amr_bit_orderings_by_mode[ctx->fr_cur_mode]);
1146
1147
    /* Decode the quantized ISF vector */
1148
6256
    if (ctx->fr_cur_mode == MODE_6k60) {
1149
512
        decode_isf_indices_36b(cf->isp_id, ctx->isf_cur);
1150
    } else {
1151
5744
        decode_isf_indices_46b(cf->isp_id, ctx->isf_cur);
1152
    }
1153
1154
6256
    isf_add_mean_and_past(ctx->isf_cur, ctx->isf_q_past);
1155
6256
    ff_set_min_dist_lsf(ctx->isf_cur, MIN_ISF_SPACING, LP_ORDER - 1);
1156
1157
6256
    stab_fac = stability_factor(ctx->isf_cur, ctx->isf_past_final);
1158
1159
6256
    ctx->isf_cur[LP_ORDER - 1] *= 2.0;
1160
6256
    ff_acelp_lsf2lspd(ctx->isp[3], ctx->isf_cur, LP_ORDER);
1161
1162
    /* Generate a ISP vector for each subframe */
1163
6256
    if (ctx->first_frame) {
1164
10
        ctx->first_frame = 0;
1165
10
        memcpy(ctx->isp_sub4_past, ctx->isp[3], LP_ORDER * sizeof(double));
1166
    }
1167
6256
    interpolate_isp(ctx->isp, ctx->isp_sub4_past);
1168
1169
31280
    for (sub = 0; sub < 4; sub++)
1170
25024
        ff_amrwb_lsp2lpc(ctx->isp[sub], ctx->lp_coef[sub], LP_ORDER);
1171
1172
31280
    for (sub = 0; sub < 4; sub++) {
1173
25024
        const AMRWBSubFrame *cur_subframe = &cf->subframe[sub];
1174
25024
        float *sub_buf = buf_out + sub * AMRWB_SFR_SIZE_16k;
1175
1176
        /* Decode adaptive codebook (pitch vector) */
1177
25024
        decode_pitch_vector(ctx, cur_subframe, sub);
1178
        /* Decode innovative codebook (fixed vector) */
1179
25024
        decode_fixed_vector(ctx->fixed_vector, cur_subframe->pul_ih,
1180
25024
                            cur_subframe->pul_il, ctx->fr_cur_mode);
1181
1182
25024
        pitch_sharpening(ctx, ctx->fixed_vector);
1183
1184
25024
        decode_gains(cur_subframe->vq_gain, ctx->fr_cur_mode,
1185
                     &fixed_gain_factor, &ctx->pitch_gain[0]);
1186
1187
25024
        ctx->fixed_gain[0] =
1188
25024
            ff_amr_set_fixed_gain(fixed_gain_factor,
1189
25024
                                  ctx->celpm_ctx.dot_productf(ctx->fixed_vector,
1190
25024
                                                               ctx->fixed_vector,
1191
                                                               AMRWB_SFR_SIZE) /
1192
                                  AMRWB_SFR_SIZE,
1193
25024
                       ctx->prediction_error,
1194
                       ENERGY_MEAN, energy_pred_fac);
1195
1196
        /* Calculate voice factor and store tilt for next subframe */
1197
25024
        voice_fac      = voice_factor(ctx->pitch_vector, ctx->pitch_gain[0],
1198
25024
                                      ctx->fixed_vector, ctx->fixed_gain[0],
1199
                                      &ctx->celpm_ctx);
1200
25024
        ctx->tilt_coef = voice_fac * 0.25 + 0.25;
1201
1202
        /* Construct current excitation */
1203
1626560
        for (i = 0; i < AMRWB_SFR_SIZE; i++) {
1204
1601536
            ctx->excitation[i] *= ctx->pitch_gain[0];
1205
1601536
            ctx->excitation[i] += ctx->fixed_gain[0] * ctx->fixed_vector[i];
1206
1601536
            ctx->excitation[i] = truncf(ctx->excitation[i]);
1207
        }
1208
1209
        /* Post-processing of excitation elements */
1210
25024
        synth_fixed_gain = noise_enhancer(ctx->fixed_gain[0], &ctx->prev_tr_gain,
1211
                                          voice_fac, stab_fac);
1212
1213
25024
        synth_fixed_vector = anti_sparseness(ctx, ctx->fixed_vector,
1214
                                             spare_vector);
1215
1216
25024
        pitch_enhancer(synth_fixed_vector, voice_fac);
1217
1218
25024
        synthesis(ctx, ctx->lp_coef[sub], synth_exc, synth_fixed_gain,
1219
                  synth_fixed_vector, &ctx->samples_az[LP_ORDER]);
1220
1221
        /* Synthesis speech post-processing */
1222
25024
        de_emphasis(&ctx->samples_up[UPS_MEM_SIZE],
1223
25024
                    &ctx->samples_az[LP_ORDER], PREEMPH_FAC, ctx->demph_mem);
1224
1225
25024
        ctx->acelpf_ctx.acelp_apply_order_2_transfer_function(&ctx->samples_up[UPS_MEM_SIZE],
1226
25024
            &ctx->samples_up[UPS_MEM_SIZE], hpf_zeros, hpf_31_poles,
1227
25024
            hpf_31_gain, ctx->hpf_31_mem, AMRWB_SFR_SIZE);
1228
1229
25024
        upsample_5_4(sub_buf, &ctx->samples_up[UPS_FIR_SIZE],
1230
                     AMRWB_SFR_SIZE_16k, &ctx->celpm_ctx);
1231
1232
        /* High frequency band (6.4 - 7.0 kHz) generation part */
1233
25024
        ctx->acelpf_ctx.acelp_apply_order_2_transfer_function(hb_samples,
1234
25024
            &ctx->samples_up[UPS_MEM_SIZE], hpf_zeros, hpf_400_poles,
1235
25024
            hpf_400_gain, ctx->hpf_400_mem, AMRWB_SFR_SIZE);
1236
1237
25024
        hb_gain = find_hb_gain(ctx, hb_samples,
1238
25024
                               cur_subframe->hb_gain, cf->vad);
1239
1240
25024
        scaled_hb_excitation(ctx, hb_exc, synth_exc, hb_gain);
1241
1242
25024
        hb_synthesis(ctx, sub, &ctx->samples_hb[LP_ORDER_16k],
1243
25024
                     hb_exc, ctx->isf_cur, ctx->isf_past_final);
1244
1245
        /* High-band post-processing filters */
1246
25024
        hb_fir_filter(hb_samples, bpf_6_7_coef, ctx->bpf_6_7_mem,
1247
25024
                      &ctx->samples_hb[LP_ORDER_16k]);
1248
1249
25024
        if (ctx->fr_cur_mode == MODE_23k85)
1250
8640
            hb_fir_filter(hb_samples, lpf_7_coef, ctx->lpf_7_mem,
1251
                          hb_samples);
1252
1253
        /* Add the low and high frequency bands */
1254
2026944
        for (i = 0; i < AMRWB_SFR_SIZE_16k; i++)
1255
2001920
            sub_buf[i] = (sub_buf[i] + hb_samples[i]) * (1.0f / (1 << 15));
1256
1257
        /* Update buffers and history */
1258
25024
        update_sub_state(ctx);
1259
    }
1260
1261
    /* update state for next frame */
1262
6256
    memcpy(ctx->isp_sub4_past, ctx->isp[3], LP_ORDER * sizeof(ctx->isp[3][0]));
1263
6256
    memcpy(ctx->isf_past_final, ctx->isf_cur, LP_ORDER * sizeof(float));
1264
1265
6256
    *got_frame_ptr = 1;
1266
1267
6256
    return expected_fr_size;
1268
}
1269
1270
AVCodec ff_amrwb_decoder = {
1271
    .name           = "amrwb",
1272
    .long_name      = NULL_IF_CONFIG_SMALL("AMR-WB (Adaptive Multi-Rate WideBand)"),
1273
    .type           = AVMEDIA_TYPE_AUDIO,
1274
    .id             = AV_CODEC_ID_AMR_WB,
1275
    .priv_data_size = sizeof(AMRWBContext),
1276
    .init           = amrwb_decode_init,
1277
    .decode         = amrwb_decode_frame,
1278
    .capabilities   = AV_CODEC_CAP_DR1,
1279
    .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT,
1280
                                                     AV_SAMPLE_FMT_NONE },
1281
};