GCC Code Coverage Report
Directory: ../../../ffmpeg/ Exec Total Coverage
File: src/libavcodec/amrnbdec.c Lines: 362 380 95.3 %
Date: 2019-11-22 03:34:36 Branches: 180 198 90.9 %

Line Branch Exec Source
1
/*
2
 * AMR narrowband decoder
3
 * Copyright (c) 2006-2007 Robert Swain
4
 * Copyright (c) 2009 Colin McQuillan
5
 *
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 * This file is part of FFmpeg.
7
 *
8
 * FFmpeg is free software; you can redistribute it and/or
9
 * modify it under the terms of the GNU Lesser General Public
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 * License as published by the Free Software Foundation; either
11
 * version 2.1 of the License, or (at your option) any later version.
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 *
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 * FFmpeg is distributed in the hope that it will be useful,
14
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
15
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
16
 * Lesser General Public License for more details.
17
 *
18
 * You should have received a copy of the GNU Lesser General Public
19
 * License along with FFmpeg; if not, write to the Free Software
20
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
21
 */
22
23
24
/**
25
 * @file
26
 * AMR narrowband decoder
27
 *
28
 * This decoder uses floats for simplicity and so is not bit-exact. One
29
 * difference is that differences in phase can accumulate. The test sequences
30
 * in 3GPP TS 26.074 can still be useful.
31
 *
32
 * - Comparing this file's output to the output of the ref decoder gives a
33
 *   PSNR of 30 to 80. Plotting the output samples shows a difference in
34
 *   phase in some areas.
35
 *
36
 * - Comparing both decoders against their input, this decoder gives a similar
37
 *   PSNR. If the test sequence homing frames are removed (this decoder does
38
 *   not detect them), the PSNR is at least as good as the reference on 140
39
 *   out of 169 tests.
40
 */
41
42
43
#include <string.h>
44
#include <math.h>
45
46
#include "libavutil/channel_layout.h"
47
#include "libavutil/float_dsp.h"
48
#include "avcodec.h"
49
#include "libavutil/common.h"
50
#include "libavutil/avassert.h"
51
#include "celp_math.h"
52
#include "celp_filters.h"
53
#include "acelp_filters.h"
54
#include "acelp_vectors.h"
55
#include "acelp_pitch_delay.h"
56
#include "lsp.h"
57
#include "amr.h"
58
#include "internal.h"
59
60
#include "amrnbdata.h"
61
62
#define AMR_BLOCK_SIZE              160   ///< samples per frame
63
#define AMR_SAMPLE_BOUND        32768.0   ///< threshold for synthesis overflow
64
65
/**
66
 * Scale from constructed speech to [-1,1]
67
 *
68
 * AMR is designed to produce 16-bit PCM samples (3GPP TS 26.090 4.2) but
69
 * upscales by two (section 6.2.2).
70
 *
71
 * Fundamentally, this scale is determined by energy_mean through
72
 * the fixed vector contribution to the excitation vector.
73
 */
74
#define AMR_SAMPLE_SCALE  (2.0 / 32768.0)
75
76
/** Prediction factor for 12.2kbit/s mode */
77
#define PRED_FAC_MODE_12k2             0.65
78
79
#define LSF_R_FAC          (8000.0 / 32768.0) ///< LSF residual tables to Hertz
80
#define MIN_LSF_SPACING    (50.0488 / 8000.0) ///< Ensures stability of LPC filter
81
#define PITCH_LAG_MIN_MODE_12k2          18   ///< Lower bound on decoded lag search in 12.2kbit/s mode
82
83
/** Initial energy in dB. Also used for bad frames (unimplemented). */
84
#define MIN_ENERGY -14.0
85
86
/** Maximum sharpening factor
87
 *
88
 * The specification says 0.8, which should be 13107, but the reference C code
89
 * uses 13017 instead. (Amusingly the same applies to SHARP_MAX in g729dec.c.)
90
 */
91
#define SHARP_MAX 0.79449462890625
92
93
/** Number of impulse response coefficients used for tilt factor */
94
#define AMR_TILT_RESPONSE   22
95
/** Tilt factor = 1st reflection coefficient * gamma_t */
96
#define AMR_TILT_GAMMA_T   0.8
97
/** Adaptive gain control factor used in post-filter */
98
#define AMR_AGC_ALPHA      0.9
99
100
typedef struct AMRContext {
101
    AMRNBFrame                        frame; ///< decoded AMR parameters (lsf coefficients, codebook indexes, etc)
102
    uint8_t             bad_frame_indicator; ///< bad frame ? 1 : 0
103
    enum Mode                cur_frame_mode;
104
105
    int16_t     prev_lsf_r[LP_FILTER_ORDER]; ///< residual LSF vector from previous subframe
106
    double          lsp[4][LP_FILTER_ORDER]; ///< lsp vectors from current frame
107
    double   prev_lsp_sub4[LP_FILTER_ORDER]; ///< lsp vector for the 4th subframe of the previous frame
108
109
    float         lsf_q[4][LP_FILTER_ORDER]; ///< Interpolated LSF vector for fixed gain smoothing
110
    float          lsf_avg[LP_FILTER_ORDER]; ///< vector of averaged lsf vector
111
112
    float           lpc[4][LP_FILTER_ORDER]; ///< lpc coefficient vectors for 4 subframes
113
114
    uint8_t                   pitch_lag_int; ///< integer part of pitch lag from current subframe
115
116
    float excitation_buf[PITCH_DELAY_MAX + LP_FILTER_ORDER + 1 + AMR_SUBFRAME_SIZE]; ///< current excitation and all necessary excitation history
117
    float                       *excitation; ///< pointer to the current excitation vector in excitation_buf
118
119
    float   pitch_vector[AMR_SUBFRAME_SIZE]; ///< adaptive code book (pitch) vector
120
    float   fixed_vector[AMR_SUBFRAME_SIZE]; ///< algebraic codebook (fixed) vector (must be kept zero between frames)
121
122
    float               prediction_error[4]; ///< quantified prediction errors {20log10(^gamma_gc)} for previous four subframes
123
    float                     pitch_gain[5]; ///< quantified pitch gains for the current and previous four subframes
124
    float                     fixed_gain[5]; ///< quantified fixed gains for the current and previous four subframes
125
126
    float                              beta; ///< previous pitch_gain, bounded by [0.0,SHARP_MAX]
127
    uint8_t                      diff_count; ///< the number of subframes for which diff has been above 0.65
128
    uint8_t                      hang_count; ///< the number of subframes since a hangover period started
129
130
    float            prev_sparse_fixed_gain; ///< previous fixed gain; used by anti-sparseness processing to determine "onset"
131
    uint8_t               prev_ir_filter_nr; ///< previous impulse response filter "impNr": 0 - strong, 1 - medium, 2 - none
132
    uint8_t                 ir_filter_onset; ///< flag for impulse response filter strength
133
134
    float                postfilter_mem[10]; ///< previous intermediate values in the formant filter
135
    float                          tilt_mem; ///< previous input to tilt compensation filter
136
    float                    postfilter_agc; ///< previous factor used for adaptive gain control
137
    float                  high_pass_mem[2]; ///< previous intermediate values in the high-pass filter
138
139
    float samples_in[LP_FILTER_ORDER + AMR_SUBFRAME_SIZE]; ///< floating point samples
140
141
    ACELPFContext                     acelpf_ctx; ///< context for filters for ACELP-based codecs
142
    ACELPVContext                     acelpv_ctx; ///< context for vector operations for ACELP-based codecs
143
    CELPFContext                       celpf_ctx; ///< context for filters for CELP-based codecs
144
    CELPMContext                       celpm_ctx; ///< context for fixed point math operations
145
146
} AMRContext;
147
148
/** Double version of ff_weighted_vector_sumf() */
149
568
static void weighted_vector_sumd(double *out, const double *in_a,
150
                                 const double *in_b, double weight_coeff_a,
151
                                 double weight_coeff_b, int length)
152
{
153
    int i;
154
155
6248
    for (i = 0; i < length; i++)
156
5680
        out[i] = weight_coeff_a * in_a[i]
157
5680
               + weight_coeff_b * in_b[i];
158
568
}
159
160
16
static av_cold int amrnb_decode_init(AVCodecContext *avctx)
161
{
162
16
    AMRContext *p = avctx->priv_data;
163
    int i;
164
165
16
    if (avctx->channels > 1) {
166
        avpriv_report_missing_feature(avctx, "multi-channel AMR");
167
        return AVERROR_PATCHWELCOME;
168
    }
169
170
16
    avctx->channels       = 1;
171
16
    avctx->channel_layout = AV_CH_LAYOUT_MONO;
172
16
    if (!avctx->sample_rate)
173
        avctx->sample_rate = 8000;
174
16
    avctx->sample_fmt     = AV_SAMPLE_FMT_FLT;
175
176
    // p->excitation always points to the same position in p->excitation_buf
177
16
    p->excitation = &p->excitation_buf[PITCH_DELAY_MAX + LP_FILTER_ORDER + 1];
178
179
176
    for (i = 0; i < LP_FILTER_ORDER; i++) {
180
160
        p->prev_lsp_sub4[i] =    lsp_sub4_init[i] * 1000 / (float)(1 << 15);
181
160
        p->lsf_avg[i] = p->lsf_q[3][i] = lsp_avg_init[i] / (float)(1 << 15);
182
    }
183
184
80
    for (i = 0; i < 4; i++)
185
64
        p->prediction_error[i] = MIN_ENERGY;
186
187
16
    ff_acelp_filter_init(&p->acelpf_ctx);
188
16
    ff_acelp_vectors_init(&p->acelpv_ctx);
189
16
    ff_celp_filter_init(&p->celpf_ctx);
190
16
    ff_celp_math_init(&p->celpm_ctx);
191
192
16
    return 0;
193
}
194
195
196
/**
197
 * Unpack an RFC4867 speech frame into the AMR frame mode and parameters.
198
 *
199
 * The order of speech bits is specified by 3GPP TS 26.101.
200
 *
201
 * @param p the context
202
 * @param buf               pointer to the input buffer
203
 * @param buf_size          size of the input buffer
204
 *
205
 * @return the frame mode
206
 */
207
2272
static enum Mode unpack_bitstream(AMRContext *p, const uint8_t *buf,
208
                                  int buf_size)
209
{
210
    enum Mode mode;
211
212
    // Decode the first octet.
213
2272
    mode = buf[0] >> 3 & 0x0F;                      // frame type
214
2272
    p->bad_frame_indicator = (buf[0] & 0x4) != 0x4; // quality bit
215
216

2272
    if (mode >= N_MODES || buf_size < frame_sizes_nb[mode] + 1) {
217
        return NO_DATA;
218
    }
219
220
2272
    if (mode < MODE_DTX)
221
2272
        ff_amr_bit_reorder((uint16_t *) &p->frame, sizeof(AMRNBFrame), buf + 1,
222
2272
                           amr_unpacking_bitmaps_per_mode[mode]);
223
224
2272
    return mode;
225
}
226
227
228
/// @name AMR pitch LPC coefficient decoding functions
229
/// @{
230
231
/**
232
 * Interpolate the LSF vector (used for fixed gain smoothing).
233
 * The interpolation is done over all four subframes even in MODE_12k2.
234
 *
235
 * @param[in]     ctx       The Context
236
 * @param[in,out] lsf_q     LSFs in [0,1] for each subframe
237
 * @param[in]     lsf_new   New LSFs in [0,1] for subframe 4
238
 */
239
2272
static void interpolate_lsf(ACELPVContext *ctx, float lsf_q[4][LP_FILTER_ORDER], float *lsf_new)
240
{
241
    int i;
242
243
11360
    for (i = 0; i < 4; i++)
244
9088
        ctx->weighted_vector_sumf(lsf_q[i], lsf_q[3], lsf_new,
245
9088
                                0.25 * (3 - i), 0.25 * (i + 1),
246
                                LP_FILTER_ORDER);
247
2272
}
248
249
/**
250
 * Decode a set of 5 split-matrix quantized lsf indexes into an lsp vector.
251
 *
252
 * @param p the context
253
 * @param lsp output LSP vector
254
 * @param lsf_no_r LSF vector without the residual vector added
255
 * @param lsf_quantizer pointers to LSF dictionary tables
256
 * @param quantizer_offset offset in tables
257
 * @param sign for the 3 dictionary table
258
 * @param update store data for computing the next frame's LSFs
259
 */
260
568
static void lsf2lsp_for_mode12k2(AMRContext *p, double lsp[LP_FILTER_ORDER],
261
                                 const float lsf_no_r[LP_FILTER_ORDER],
262
                                 const int16_t *lsf_quantizer[5],
263
                                 const int quantizer_offset,
264
                                 const int sign, const int update)
265
{
266
    int16_t lsf_r[LP_FILTER_ORDER]; // residual LSF vector
267
    float lsf_q[LP_FILTER_ORDER]; // quantified LSF vector
268
    int i;
269
270
3408
    for (i = 0; i < LP_FILTER_ORDER >> 1; i++)
271
2840
        memcpy(&lsf_r[i << 1], &lsf_quantizer[i][quantizer_offset],
272
               2 * sizeof(*lsf_r));
273
274
568
    if (sign) {
275
266
        lsf_r[4] *= -1;
276
266
        lsf_r[5] *= -1;
277
    }
278
279
568
    if (update)
280
284
        memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(*lsf_r));
281
282
6248
    for (i = 0; i < LP_FILTER_ORDER; i++)
283
5680
        lsf_q[i] = lsf_r[i] * (LSF_R_FAC / 8000.0) + lsf_no_r[i] * (1.0 / 8000.0);
284
285
568
    ff_set_min_dist_lsf(lsf_q, MIN_LSF_SPACING, LP_FILTER_ORDER);
286
287
568
    if (update)
288
284
        interpolate_lsf(&p->acelpv_ctx, p->lsf_q, lsf_q);
289
290
568
    ff_acelp_lsf2lspd(lsp, lsf_q, LP_FILTER_ORDER);
291
568
}
292
293
/**
294
 * Decode a set of 5 split-matrix quantized lsf indexes into 2 lsp vectors.
295
 *
296
 * @param p                 pointer to the AMRContext
297
 */
298
284
static void lsf2lsp_5(AMRContext *p)
299
{
300
284
    const uint16_t *lsf_param = p->frame.lsf;
301
    float lsf_no_r[LP_FILTER_ORDER]; // LSFs without the residual vector
302
    const int16_t *lsf_quantizer[5];
303
    int i;
304
305
284
    lsf_quantizer[0] = lsf_5_1[lsf_param[0]];
306
284
    lsf_quantizer[1] = lsf_5_2[lsf_param[1]];
307
284
    lsf_quantizer[2] = lsf_5_3[lsf_param[2] >> 1];
308
284
    lsf_quantizer[3] = lsf_5_4[lsf_param[3]];
309
284
    lsf_quantizer[4] = lsf_5_5[lsf_param[4]];
310
311
3124
    for (i = 0; i < LP_FILTER_ORDER; i++)
312
2840
        lsf_no_r[i] = p->prev_lsf_r[i] * LSF_R_FAC * PRED_FAC_MODE_12k2 + lsf_5_mean[i];
313
314
284
    lsf2lsp_for_mode12k2(p, p->lsp[1], lsf_no_r, lsf_quantizer, 0, lsf_param[2] & 1, 0);
315
284
    lsf2lsp_for_mode12k2(p, p->lsp[3], lsf_no_r, lsf_quantizer, 2, lsf_param[2] & 1, 1);
316
317
    // interpolate LSP vectors at subframes 1 and 3
318
284
    weighted_vector_sumd(p->lsp[0], p->prev_lsp_sub4, p->lsp[1], 0.5, 0.5, LP_FILTER_ORDER);
319
284
    weighted_vector_sumd(p->lsp[2], p->lsp[1]       , p->lsp[3], 0.5, 0.5, LP_FILTER_ORDER);
320
284
}
321
322
/**
323
 * Decode a set of 3 split-matrix quantized lsf indexes into an lsp vector.
324
 *
325
 * @param p                 pointer to the AMRContext
326
 */
327
1988
static void lsf2lsp_3(AMRContext *p)
328
{
329
1988
    const uint16_t *lsf_param = p->frame.lsf;
330
    int16_t lsf_r[LP_FILTER_ORDER]; // residual LSF vector
331
    float lsf_q[LP_FILTER_ORDER]; // quantified LSF vector
332
    const int16_t *lsf_quantizer;
333
    int i, j;
334
335
1988
    lsf_quantizer = (p->cur_frame_mode == MODE_7k95 ? lsf_3_1_MODE_7k95 : lsf_3_1)[lsf_param[0]];
336
1988
    memcpy(lsf_r, lsf_quantizer, 3 * sizeof(*lsf_r));
337
338
1988
    lsf_quantizer = lsf_3_2[lsf_param[1] << (p->cur_frame_mode <= MODE_5k15)];
339
1988
    memcpy(lsf_r + 3, lsf_quantizer, 3 * sizeof(*lsf_r));
340
341
1988
    lsf_quantizer = (p->cur_frame_mode <= MODE_5k15 ? lsf_3_3_MODE_5k15 : lsf_3_3)[lsf_param[2]];
342
1988
    memcpy(lsf_r + 6, lsf_quantizer, 4 * sizeof(*lsf_r));
343
344
    // calculate mean-removed LSF vector and add mean
345
21868
    for (i = 0; i < LP_FILTER_ORDER; i++)
346
19880
        lsf_q[i] = (lsf_r[i] + p->prev_lsf_r[i] * pred_fac[i]) * (LSF_R_FAC / 8000.0) + lsf_3_mean[i] * (1.0 / 8000.0);
347
348
1988
    ff_set_min_dist_lsf(lsf_q, MIN_LSF_SPACING, LP_FILTER_ORDER);
349
350
    // store data for computing the next frame's LSFs
351
1988
    interpolate_lsf(&p->acelpv_ctx, p->lsf_q, lsf_q);
352
1988
    memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(*lsf_r));
353
354
1988
    ff_acelp_lsf2lspd(p->lsp[3], lsf_q, LP_FILTER_ORDER);
355
356
    // interpolate LSP vectors at subframes 1, 2 and 3
357
7952
    for (i = 1; i <= 3; i++)
358
65604
        for(j = 0; j < LP_FILTER_ORDER; j++)
359
59640
            p->lsp[i-1][j] = p->prev_lsp_sub4[j] +
360
59640
                (p->lsp[3][j] - p->prev_lsp_sub4[j]) * 0.25 * i;
361
1988
}
362
363
/// @}
364
365
366
/// @name AMR pitch vector decoding functions
367
/// @{
368
369
/**
370
 * Like ff_decode_pitch_lag(), but with 1/6 resolution
371
 */
372
1136
static void decode_pitch_lag_1_6(int *lag_int, int *lag_frac, int pitch_index,
373
                                 const int prev_lag_int, const int subframe)
374
{
375

1136
    if (subframe == 0 || subframe == 2) {
376
568
        if (pitch_index < 463) {
377
457
            *lag_int  = (pitch_index + 107) * 10923 >> 16;
378
457
            *lag_frac = pitch_index - *lag_int * 6 + 105;
379
        } else {
380
111
            *lag_int  = pitch_index - 368;
381
111
            *lag_frac = 0;
382
        }
383
    } else {
384
568
        *lag_int  = ((pitch_index + 5) * 10923 >> 16) - 1;
385
568
        *lag_frac = pitch_index - *lag_int * 6 - 3;
386
568
        *lag_int += av_clip(prev_lag_int - 5, PITCH_LAG_MIN_MODE_12k2,
387
                            PITCH_DELAY_MAX - 9);
388
    }
389
1136
}
390
391
9088
static void decode_pitch_vector(AMRContext *p,
392
                                const AMRNBSubframe *amr_subframe,
393
                                const int subframe)
394
{
395
    int pitch_lag_int, pitch_lag_frac;
396
9088
    enum Mode mode = p->cur_frame_mode;
397
398
9088
    if (p->cur_frame_mode == MODE_12k2) {
399
1136
        decode_pitch_lag_1_6(&pitch_lag_int, &pitch_lag_frac,
400
1136
                             amr_subframe->p_lag, p->pitch_lag_int,
401
                             subframe);
402
    } else {
403

11360
        ff_decode_pitch_lag(&pitch_lag_int, &pitch_lag_frac,
404
7952
                            amr_subframe->p_lag,
405
7952
                            p->pitch_lag_int, subframe,
406
                            mode != MODE_4k75 && mode != MODE_5k15,
407
3408
                            mode <= MODE_6k7 ? 4 : (mode == MODE_7k95 ? 5 : 6));
408
7952
        pitch_lag_frac *= 2;
409
    }
410
411
9088
    p->pitch_lag_int = pitch_lag_int; // store previous lag in a uint8_t
412
413
9088
    pitch_lag_int += pitch_lag_frac > 0;
414
415
    /* Calculate the pitch vector by interpolating the past excitation at the
416
       pitch lag using a b60 hamming windowed sinc function.   */
417
18176
    p->acelpf_ctx.acelp_interpolatef(p->excitation,
418
9088
                          p->excitation + 1 - pitch_lag_int,
419
                          ff_b60_sinc, 6,
420
9088
                          pitch_lag_frac + 6 - 6*(pitch_lag_frac > 0),
421
                          10, AMR_SUBFRAME_SIZE);
422
423
9088
    memcpy(p->pitch_vector, p->excitation, AMR_SUBFRAME_SIZE * sizeof(float));
424
9088
}
425
426
/// @}
427
428
429
/// @name AMR algebraic code book (fixed) vector decoding functions
430
/// @{
431
432
/**
433
 * Decode a 10-bit algebraic codebook index from a 10.2 kbit/s frame.
434
 */
435
2272
static void decode_10bit_pulse(int code, int pulse_position[8],
436
                               int i1, int i2, int i3)
437
{
438
    // coded using 7+3 bits with the 3 LSBs being, individually, the LSB of 1 of
439
    // the 3 pulses and the upper 7 bits being coded in base 5
440
2272
    const uint8_t *positions = base_five_table[code >> 3];
441
2272
    pulse_position[i1] = (positions[2] << 1) + ( code       & 1);
442
2272
    pulse_position[i2] = (positions[1] << 1) + ((code >> 1) & 1);
443
2272
    pulse_position[i3] = (positions[0] << 1) + ((code >> 2) & 1);
444
2272
}
445
446
/**
447
 * Decode the algebraic codebook index to pulse positions and signs and
448
 * construct the algebraic codebook vector for MODE_10k2.
449
 *
450
 * @param fixed_index          positions of the eight pulses
451
 * @param fixed_sparse         pointer to the algebraic codebook vector
452
 */
453
1136
static void decode_8_pulses_31bits(const int16_t *fixed_index,
454
                                   AMRFixed *fixed_sparse)
455
{
456
    int pulse_position[8];
457
    int i, temp;
458
459
1136
    decode_10bit_pulse(fixed_index[4], pulse_position, 0, 4, 1);
460
1136
    decode_10bit_pulse(fixed_index[5], pulse_position, 2, 6, 5);
461
462
    // coded using 5+2 bits with the 2 LSBs being, individually, the LSB of 1 of
463
    // the 2 pulses and the upper 5 bits being coded in base 5
464
1136
    temp = ((fixed_index[6] >> 2) * 25 + 12) >> 5;
465
1136
    pulse_position[3] = temp % 5;
466
1136
    pulse_position[7] = temp / 5;
467
1136
    if (pulse_position[7] & 1)
468
477
        pulse_position[3] = 4 - pulse_position[3];
469
1136
    pulse_position[3] = (pulse_position[3] << 1) + ( fixed_index[6]       & 1);
470
1136
    pulse_position[7] = (pulse_position[7] << 1) + ((fixed_index[6] >> 1) & 1);
471
472
1136
    fixed_sparse->n = 8;
473
5680
    for (i = 0; i < 4; i++) {
474
4544
        const int pos1   = (pulse_position[i]     << 2) + i;
475
4544
        const int pos2   = (pulse_position[i + 4] << 2) + i;
476
4544
        const float sign = fixed_index[i] ? -1.0 : 1.0;
477
4544
        fixed_sparse->x[i    ] = pos1;
478
4544
        fixed_sparse->x[i + 4] = pos2;
479
4544
        fixed_sparse->y[i    ] = sign;
480
4544
        fixed_sparse->y[i + 4] = pos2 < pos1 ? -sign : sign;
481
    }
482
1136
}
483
484
/**
485
 * Decode the algebraic codebook index to pulse positions and signs,
486
 * then construct the algebraic codebook vector.
487
 *
488
 *                              nb of pulses | bits encoding pulses
489
 * For MODE_4k75 or MODE_5k15,             2 | 1-3, 4-6, 7
490
 *                  MODE_5k9,              2 | 1,   2-4, 5-6, 7-9
491
 *                  MODE_6k7,              3 | 1-3, 4,   5-7, 8,  9-11
492
 *      MODE_7k4 or MODE_7k95,             4 | 1-3, 4-6, 7-9, 10, 11-13
493
 *
494
 * @param fixed_sparse pointer to the algebraic codebook vector
495
 * @param pulses       algebraic codebook indexes
496
 * @param mode         mode of the current frame
497
 * @param subframe     current subframe number
498
 */
499
9088
static void decode_fixed_sparse(AMRFixed *fixed_sparse, const uint16_t *pulses,
500
                                const enum Mode mode, const int subframe)
501
{
502
    av_assert1(MODE_4k75 <= (signed)mode && mode <= MODE_12k2);
503
504
9088
    if (mode == MODE_12k2) {
505
1136
        ff_decode_10_pulses_35bits(pulses, fixed_sparse, gray_decode, 5, 3);
506
7952
    } else if (mode == MODE_10k2) {
507
1136
        decode_8_pulses_31bits(pulses, fixed_sparse);
508
    } else {
509
6816
        int *pulse_position = fixed_sparse->x;
510
        int i, pulse_subset;
511
6816
        const int fixed_index = pulses[0];
512
513
6816
        if (mode <= MODE_5k15) {
514
2272
            pulse_subset      = ((fixed_index >> 3) & 8)     + (subframe << 1);
515
2272
            pulse_position[0] = ( fixed_index       & 7) * 5 + track_position[pulse_subset];
516
2272
            pulse_position[1] = ((fixed_index >> 3) & 7) * 5 + track_position[pulse_subset + 1];
517
2272
            fixed_sparse->n = 2;
518
4544
        } else if (mode == MODE_5k9) {
519
1136
            pulse_subset      = ((fixed_index & 1) << 1) + 1;
520
1136
            pulse_position[0] = ((fixed_index >> 1) & 7) * 5 + pulse_subset;
521
1136
            pulse_subset      = (fixed_index  >> 4) & 3;
522
1136
            pulse_position[1] = ((fixed_index >> 6) & 7) * 5 + pulse_subset + (pulse_subset == 3 ? 1 : 0);
523
1136
            fixed_sparse->n = pulse_position[0] == pulse_position[1] ? 1 : 2;
524
3408
        } else if (mode == MODE_6k7) {
525
1136
            pulse_position[0] = (fixed_index        & 7) * 5;
526
1136
            pulse_subset      = (fixed_index  >> 2) & 2;
527
1136
            pulse_position[1] = ((fixed_index >> 4) & 7) * 5 + pulse_subset + 1;
528
1136
            pulse_subset      = (fixed_index  >> 6) & 2;
529
1136
            pulse_position[2] = ((fixed_index >> 8) & 7) * 5 + pulse_subset + 2;
530
1136
            fixed_sparse->n = 3;
531
        } else { // mode <= MODE_7k95
532
2272
            pulse_position[0] = gray_decode[ fixed_index        & 7];
533
2272
            pulse_position[1] = gray_decode[(fixed_index >> 3)  & 7] + 1;
534
2272
            pulse_position[2] = gray_decode[(fixed_index >> 6)  & 7] + 2;
535
2272
            pulse_subset      = (fixed_index >> 9) & 1;
536
2272
            pulse_position[3] = gray_decode[(fixed_index >> 10) & 7] + pulse_subset + 3;
537
2272
            fixed_sparse->n = 4;
538
        }
539
26127
        for (i = 0; i < fixed_sparse->n; i++)
540
19311
            fixed_sparse->y[i] = (pulses[1] >> i) & 1 ? 1.0 : -1.0;
541
    }
542
9088
}
543
544
/**
545
 * Apply pitch lag to obtain the sharpened fixed vector (section 6.1.2)
546
 *
547
 * @param p the context
548
 * @param subframe unpacked amr subframe
549
 * @param mode mode of the current frame
550
 * @param fixed_sparse sparse representation of the fixed vector
551
 */
552
9088
static void pitch_sharpening(AMRContext *p, int subframe, enum Mode mode,
553
                             AMRFixed *fixed_sparse)
554
{
555
    // The spec suggests the current pitch gain is always used, but in other
556
    // modes the pitch and codebook gains are jointly quantized (sec 5.8.2)
557
    // so the codebook gain cannot depend on the quantized pitch gain.
558
9088
    if (mode == MODE_12k2)
559
1136
        p->beta = FFMIN(p->pitch_gain[4], 1.0);
560
561
9088
    fixed_sparse->pitch_lag  = p->pitch_lag_int;
562
9088
    fixed_sparse->pitch_fac  = p->beta;
563
564
    // Save pitch sharpening factor for the next subframe
565
    // MODE_4k75 only updates on the 2nd and 4th subframes - this follows from
566
    // the fact that the gains for two subframes are jointly quantized.
567

9088
    if (mode != MODE_4k75 || subframe & 1)
568
8520
        p->beta = av_clipf(p->pitch_gain[4], 0.0, SHARP_MAX);
569
9088
}
570
/// @}
571
572
573
/// @name AMR gain decoding functions
574
/// @{
575
576
/**
577
 * fixed gain smoothing
578
 * Note that where the spec specifies the "spectrum in the q domain"
579
 * in section 6.1.4, in fact frequencies should be used.
580
 *
581
 * @param p the context
582
 * @param lsf LSFs for the current subframe, in the range [0,1]
583
 * @param lsf_avg averaged LSFs
584
 * @param mode mode of the current frame
585
 *
586
 * @return fixed gain smoothed
587
 */
588
9088
static float fixed_gain_smooth(AMRContext *p , const float *lsf,
589
                               const float *lsf_avg, const enum Mode mode)
590
{
591
9088
    float diff = 0.0;
592
    int i;
593
594
99968
    for (i = 0; i < LP_FILTER_ORDER; i++)
595
90880
        diff += fabs(lsf_avg[i] - lsf[i]) / lsf_avg[i];
596
597
    // If diff is large for ten subframes, disable smoothing for a 40-subframe
598
    // hangover period.
599
9088
    p->diff_count++;
600
9088
    if (diff <= 0.65)
601
6688
        p->diff_count = 0;
602
603
9088
    if (p->diff_count > 10) {
604
687
        p->hang_count = 0;
605
687
        p->diff_count--; // don't let diff_count overflow
606
    }
607
608
9088
    if (p->hang_count < 40) {
609
3803
        p->hang_count++;
610

5285
    } else if (mode < MODE_7k4 || mode == MODE_10k2) {
611
3413
        const float smoothing_factor = av_clipf(4.0 * diff - 1.6, 0.0, 1.0);
612
3413
        const float fixed_gain_mean = (p->fixed_gain[0] + p->fixed_gain[1] +
613
3413
                                       p->fixed_gain[2] + p->fixed_gain[3] +
614
3413
                                       p->fixed_gain[4]) * 0.2;
615
3413
        return smoothing_factor * p->fixed_gain[4] +
616
3413
               (1.0 - smoothing_factor) * fixed_gain_mean;
617
    }
618
5675
    return p->fixed_gain[4];
619
}
620
621
/**
622
 * Decode pitch gain and fixed gain factor (part of section 6.1.3).
623
 *
624
 * @param p the context
625
 * @param amr_subframe unpacked amr subframe
626
 * @param mode mode of the current frame
627
 * @param subframe current subframe number
628
 * @param fixed_gain_factor decoded gain correction factor
629
 */
630
9088
static void decode_gains(AMRContext *p, const AMRNBSubframe *amr_subframe,
631
                         const enum Mode mode, const int subframe,
632
                         float *fixed_gain_factor)
633
{
634

9088
    if (mode == MODE_12k2 || mode == MODE_7k95) {
635
2272
        p->pitch_gain[4]   = qua_gain_pit [amr_subframe->p_gain    ]
636
2272
            * (1.0 / 16384.0);
637
2272
        *fixed_gain_factor = qua_gain_code[amr_subframe->fixed_gain]
638
2272
            * (1.0 /  2048.0);
639
    } else {
640
        const uint16_t *gains;
641
642
6816
        if (mode >= MODE_6k7) {
643
3408
            gains = gains_high[amr_subframe->p_gain];
644
3408
        } else if (mode >= MODE_5k15) {
645
2272
            gains = gains_low [amr_subframe->p_gain];
646
        } else {
647
            // gain index is only coded in subframes 0,2 for MODE_4k75
648
1136
            gains = gains_MODE_4k75[(p->frame.subframe[subframe & 2].p_gain << 1) + (subframe & 1)];
649
        }
650
651
6816
        p->pitch_gain[4]   = gains[0] * (1.0 / 16384.0);
652
6816
        *fixed_gain_factor = gains[1] * (1.0 /  4096.0);
653
    }
654
9088
}
655
656
/// @}
657
658
659
/// @name AMR preprocessing functions
660
/// @{
661
662
/**
663
 * Circularly convolve a sparse fixed vector with a phase dispersion impulse
664
 * response filter (D.6.2 of G.729 and 6.1.5 of AMR).
665
 *
666
 * @param out vector with filter applied
667
 * @param in source vector
668
 * @param filter phase filter coefficients
669
 *
670
 *  out[n] = sum(i,0,len-1){ in[i] * filter[(len + n - i)%len] }
671
 */
672
4859
static void apply_ir_filter(float *out, const AMRFixed *in,
673
                            const float *filter)
674
{
675
    float filter1[AMR_SUBFRAME_SIZE],     ///< filters at pitch lag*1 and *2
676
          filter2[AMR_SUBFRAME_SIZE];
677
4859
    int   lag = in->pitch_lag;
678
4859
    float fac = in->pitch_fac;
679
    int i;
680
681
4859
    if (lag < AMR_SUBFRAME_SIZE) {
682
731
        ff_celp_circ_addf(filter1, filter, filter, lag, fac,
683
                          AMR_SUBFRAME_SIZE);
684
685
731
        if (lag < AMR_SUBFRAME_SIZE >> 1)
686
            ff_celp_circ_addf(filter2, filter, filter1, lag, fac,
687
                              AMR_SUBFRAME_SIZE);
688
    }
689
690
4859
    memset(out, 0, sizeof(float) * AMR_SUBFRAME_SIZE);
691
17641
    for (i = 0; i < in->n; i++) {
692
12782
        int   x = in->x[i];
693
12782
        float y = in->y[i];
694
        const float *filterp;
695
696
12782
        if (x >= AMR_SUBFRAME_SIZE - lag) {
697
12467
            filterp = filter;
698
315
        } else if (x >= AMR_SUBFRAME_SIZE - (lag << 1)) {
699
315
            filterp = filter1;
700
        } else
701
            filterp = filter2;
702
703
12782
        ff_celp_circ_addf(out, out, filterp, x, y, AMR_SUBFRAME_SIZE);
704
    }
705
4859
}
706
707
/**
708
 * Reduce fixed vector sparseness by smoothing with one of three IR filters.
709
 * Also know as "adaptive phase dispersion".
710
 *
711
 * This implements 3GPP TS 26.090 section 6.1(5).
712
 *
713
 * @param p the context
714
 * @param fixed_sparse algebraic codebook vector
715
 * @param fixed_vector unfiltered fixed vector
716
 * @param fixed_gain smoothed gain
717
 * @param out space for modified vector if necessary
718
 */
719
9088
static const float *anti_sparseness(AMRContext *p, AMRFixed *fixed_sparse,
720
                                    const float *fixed_vector,
721
                                    float fixed_gain, float *out)
722
{
723
    int ir_filter_nr;
724
725
9088
    if (p->pitch_gain[4] < 0.6) {
726
4143
        ir_filter_nr = 0;      // strong filtering
727
4945
    } else if (p->pitch_gain[4] < 0.9) {
728
2985
        ir_filter_nr = 1;      // medium filtering
729
    } else
730
1960
        ir_filter_nr = 2;      // no filtering
731
732
    // detect 'onset'
733
9088
    if (fixed_gain > 2.0 * p->prev_sparse_fixed_gain) {
734
189
        p->ir_filter_onset = 2;
735
8899
    } else if (p->ir_filter_onset)
736
331
        p->ir_filter_onset--;
737
738
9088
    if (!p->ir_filter_onset) {
739
8732
        int i, count = 0;
740
741
52392
        for (i = 0; i < 5; i++)
742
43660
            if (p->pitch_gain[i] < 0.6)
743
19971
                count++;
744
8732
        if (count > 2)
745
3912
            ir_filter_nr = 0;
746
747
8732
        if (ir_filter_nr > p->prev_ir_filter_nr + 1)
748
489
            ir_filter_nr--;
749
356
    } else if (ir_filter_nr < 2)
750
230
        ir_filter_nr++;
751
752
    // Disable filtering for very low level of fixed_gain.
753
    // Note this step is not specified in the technical description but is in
754
    // the reference source in the function Ph_disp.
755
9088
    if (fixed_gain < 5.0)
756
7
        ir_filter_nr = 2;
757
758

9088
    if (p->cur_frame_mode != MODE_7k4 && p->cur_frame_mode < MODE_10k2
759
5680
         && ir_filter_nr < 2) {
760
4859
        apply_ir_filter(out, fixed_sparse,
761
4859
                        (p->cur_frame_mode == MODE_7k95 ?
762
4859
                             ir_filters_lookup_MODE_7k95 :
763
4859
                             ir_filters_lookup)[ir_filter_nr]);
764
4859
        fixed_vector = out;
765
    }
766
767
    // update ir filter strength history
768
9088
    p->prev_ir_filter_nr       = ir_filter_nr;
769
9088
    p->prev_sparse_fixed_gain  = fixed_gain;
770
771
9088
    return fixed_vector;
772
}
773
774
/// @}
775
776
777
/// @name AMR synthesis functions
778
/// @{
779
780
/**
781
 * Conduct 10th order linear predictive coding synthesis.
782
 *
783
 * @param p             pointer to the AMRContext
784
 * @param lpc           pointer to the LPC coefficients
785
 * @param fixed_gain    fixed codebook gain for synthesis
786
 * @param fixed_vector  algebraic codebook vector
787
 * @param samples       pointer to the output speech samples
788
 * @param overflow      16-bit overflow flag
789
 */
790
9088
static int synthesis(AMRContext *p, float *lpc,
791
                     float fixed_gain, const float *fixed_vector,
792
                     float *samples, uint8_t overflow)
793
{
794
    int i;
795
    float excitation[AMR_SUBFRAME_SIZE];
796
797
    // if an overflow has been detected, the pitch vector is scaled down by a
798
    // factor of 4
799
9088
    if (overflow)
800
        for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
801
            p->pitch_vector[i] *= 0.25;
802
803
9088
    p->acelpv_ctx.weighted_vector_sumf(excitation, p->pitch_vector, fixed_vector,
804
                            p->pitch_gain[4], fixed_gain, AMR_SUBFRAME_SIZE);
805
806
    // emphasize pitch vector contribution
807

9088
    if (p->pitch_gain[4] > 0.5 && !overflow) {
808
6201
        float energy = p->celpm_ctx.dot_productf(excitation, excitation,
809
                                                    AMR_SUBFRAME_SIZE);
810
6201
        float pitch_factor =
811
12402
            p->pitch_gain[4] *
812
6201
            (p->cur_frame_mode == MODE_12k2 ?
813

11700
                0.25 * FFMIN(p->pitch_gain[4], 1.0) :
814
5499
                0.5  * FFMIN(p->pitch_gain[4], SHARP_MAX));
815
816
254241
        for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
817
248040
            excitation[i] += pitch_factor * p->pitch_vector[i];
818
819
6201
        ff_scale_vector_to_given_sum_of_squares(excitation, excitation, energy,
820
                                                AMR_SUBFRAME_SIZE);
821
    }
822
823
9088
    p->celpf_ctx.celp_lp_synthesis_filterf(samples, lpc, excitation,
824
                                 AMR_SUBFRAME_SIZE,
825
                                 LP_FILTER_ORDER);
826
827
    // detect overflow
828
372608
    for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
829
363520
        if (fabsf(samples[i]) > AMR_SAMPLE_BOUND) {
830
            return 1;
831
        }
832
833
9088
    return 0;
834
}
835
836
/// @}
837
838
839
/// @name AMR update functions
840
/// @{
841
842
/**
843
 * Update buffers and history at the end of decoding a subframe.
844
 *
845
 * @param p             pointer to the AMRContext
846
 */
847
9088
static void update_state(AMRContext *p)
848
{
849
9088
    memcpy(p->prev_lsp_sub4, p->lsp[3], LP_FILTER_ORDER * sizeof(p->lsp[3][0]));
850
851
9088
    memmove(&p->excitation_buf[0], &p->excitation_buf[AMR_SUBFRAME_SIZE],
852
            (PITCH_DELAY_MAX + LP_FILTER_ORDER + 1) * sizeof(float));
853
854
9088
    memmove(&p->pitch_gain[0], &p->pitch_gain[1], 4 * sizeof(float));
855
9088
    memmove(&p->fixed_gain[0], &p->fixed_gain[1], 4 * sizeof(float));
856
857
9088
    memmove(&p->samples_in[0], &p->samples_in[AMR_SUBFRAME_SIZE],
858
            LP_FILTER_ORDER * sizeof(float));
859
9088
}
860
861
/// @}
862
863
864
/// @name AMR Postprocessing functions
865
/// @{
866
867
/**
868
 * Get the tilt factor of a formant filter from its transfer function
869
 *
870
 * @param p     The Context
871
 * @param lpc_n LP_FILTER_ORDER coefficients of the numerator
872
 * @param lpc_d LP_FILTER_ORDER coefficients of the denominator
873
 */
874
9088
static float tilt_factor(AMRContext *p, float *lpc_n, float *lpc_d)
875
{
876
    float rh0, rh1; // autocorrelation at lag 0 and 1
877
878
    // LP_FILTER_ORDER prior zeros are needed for ff_celp_lp_synthesis_filterf
879
9088
    float impulse_buffer[LP_FILTER_ORDER + AMR_TILT_RESPONSE] = { 0 };
880
9088
    float *hf = impulse_buffer + LP_FILTER_ORDER; // start of impulse response
881
882
9088
    hf[0] = 1.0;
883
9088
    memcpy(hf + 1, lpc_n, sizeof(float) * LP_FILTER_ORDER);
884
9088
    p->celpf_ctx.celp_lp_synthesis_filterf(hf, lpc_d, hf,
885
                                 AMR_TILT_RESPONSE,
886
                                 LP_FILTER_ORDER);
887
888
9088
    rh0 = p->celpm_ctx.dot_productf(hf, hf,     AMR_TILT_RESPONSE);
889
9088
    rh1 = p->celpm_ctx.dot_productf(hf, hf + 1, AMR_TILT_RESPONSE - 1);
890
891
    // The spec only specifies this check for 12.2 and 10.2 kbit/s
892
    // modes. But in the ref source the tilt is always non-negative.
893
9088
    return rh1 >= 0.0 ? rh1 / rh0 * AMR_TILT_GAMMA_T : 0.0;
894
}
895
896
/**
897
 * Perform adaptive post-filtering to enhance the quality of the speech.
898
 * See section 6.2.1.
899
 *
900
 * @param p             pointer to the AMRContext
901
 * @param lpc           interpolated LP coefficients for this subframe
902
 * @param buf_out       output of the filter
903
 */
904
9088
static void postfilter(AMRContext *p, float *lpc, float *buf_out)
905
{
906
    int i;
907
9088
    float *samples          = p->samples_in + LP_FILTER_ORDER; // Start of input
908
909
9088
    float speech_gain       = p->celpm_ctx.dot_productf(samples, samples,
910
                                                           AMR_SUBFRAME_SIZE);
911
912
    float pole_out[AMR_SUBFRAME_SIZE + LP_FILTER_ORDER];  // Output of pole filter
913
    const float *gamma_n, *gamma_d;                       // Formant filter factor table
914
    float lpc_n[LP_FILTER_ORDER], lpc_d[LP_FILTER_ORDER]; // Transfer function coefficients
915
916

9088
    if (p->cur_frame_mode == MODE_12k2 || p->cur_frame_mode == MODE_10k2) {
917
2272
        gamma_n = ff_pow_0_7;
918
2272
        gamma_d = ff_pow_0_75;
919
    } else {
920
6816
        gamma_n = ff_pow_0_55;
921
6816
        gamma_d = ff_pow_0_7;
922
    }
923
924
99968
    for (i = 0; i < LP_FILTER_ORDER; i++) {
925
90880
         lpc_n[i] = lpc[i] * gamma_n[i];
926
90880
         lpc_d[i] = lpc[i] * gamma_d[i];
927
    }
928
929
9088
    memcpy(pole_out, p->postfilter_mem, sizeof(float) * LP_FILTER_ORDER);
930
9088
    p->celpf_ctx.celp_lp_synthesis_filterf(pole_out + LP_FILTER_ORDER, lpc_d, samples,
931
                                 AMR_SUBFRAME_SIZE, LP_FILTER_ORDER);
932
9088
    memcpy(p->postfilter_mem, pole_out + AMR_SUBFRAME_SIZE,
933
           sizeof(float) * LP_FILTER_ORDER);
934
935
9088
    p->celpf_ctx.celp_lp_zero_synthesis_filterf(buf_out, lpc_n,
936
                                      pole_out + LP_FILTER_ORDER,
937
                                      AMR_SUBFRAME_SIZE, LP_FILTER_ORDER);
938
939
9088
    ff_tilt_compensation(&p->tilt_mem, tilt_factor(p, lpc_n, lpc_d), buf_out,
940
                         AMR_SUBFRAME_SIZE);
941
942
9088
    ff_adaptive_gain_control(buf_out, buf_out, speech_gain, AMR_SUBFRAME_SIZE,
943
                             AMR_AGC_ALPHA, &p->postfilter_agc);
944
9088
}
945
946
/// @}
947
948
2272
static int amrnb_decode_frame(AVCodecContext *avctx, void *data,
949
                              int *got_frame_ptr, AVPacket *avpkt)
950
{
951
952
2272
    AMRContext *p = avctx->priv_data;        // pointer to private data
953
2272
    AVFrame *frame     = data;
954
2272
    const uint8_t *buf = avpkt->data;
955
2272
    int buf_size       = avpkt->size;
956
    float *buf_out;                          // pointer to the output data buffer
957
    int i, subframe, ret;
958
    float fixed_gain_factor;
959
2272
    AMRFixed fixed_sparse = {0};             // fixed vector up to anti-sparseness processing
960
    float spare_vector[AMR_SUBFRAME_SIZE];   // extra stack space to hold result from anti-sparseness processing
961
    float synth_fixed_gain;                  // the fixed gain that synthesis should use
962
    const float *synth_fixed_vector;         // pointer to the fixed vector that synthesis should use
963
964
    /* get output buffer */
965
2272
    frame->nb_samples = AMR_BLOCK_SIZE;
966
2272
    if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
967
        return ret;
968
2272
    buf_out = (float *)frame->data[0];
969
970
2272
    p->cur_frame_mode = unpack_bitstream(p, buf, buf_size);
971
2272
    if (p->cur_frame_mode == NO_DATA) {
972
        av_log(avctx, AV_LOG_ERROR, "Corrupt bitstream\n");
973
        return AVERROR_INVALIDDATA;
974
    }
975
2272
    if (p->cur_frame_mode == MODE_DTX) {
976
        avpriv_report_missing_feature(avctx, "dtx mode");
977
        av_log(avctx, AV_LOG_INFO, "Note: libopencore_amrnb supports dtx\n");
978
        return AVERROR_PATCHWELCOME;
979
    }
980
981
2272
    if (p->cur_frame_mode == MODE_12k2) {
982
284
        lsf2lsp_5(p);
983
    } else
984
1988
        lsf2lsp_3(p);
985
986
11360
    for (i = 0; i < 4; i++)
987
9088
        ff_acelp_lspd2lpc(p->lsp[i], p->lpc[i], 5);
988
989
11360
    for (subframe = 0; subframe < 4; subframe++) {
990
9088
        const AMRNBSubframe *amr_subframe = &p->frame.subframe[subframe];
991
992
9088
        decode_pitch_vector(p, amr_subframe, subframe);
993
994
9088
        decode_fixed_sparse(&fixed_sparse, amr_subframe->pulses,
995
                            p->cur_frame_mode, subframe);
996
997
        // The fixed gain (section 6.1.3) depends on the fixed vector
998
        // (section 6.1.2), but the fixed vector calculation uses
999
        // pitch sharpening based on the on the pitch gain (section 6.1.3).
1000
        // So the correct order is: pitch gain, pitch sharpening, fixed gain.
1001
9088
        decode_gains(p, amr_subframe, p->cur_frame_mode, subframe,
1002
                     &fixed_gain_factor);
1003
1004
9088
        pitch_sharpening(p, subframe, p->cur_frame_mode, &fixed_sparse);
1005
1006
9088
        if (fixed_sparse.pitch_lag == 0) {
1007
            av_log(avctx, AV_LOG_ERROR, "The file is corrupted, pitch_lag = 0 is not allowed\n");
1008
            return AVERROR_INVALIDDATA;
1009
        }
1010
9088
        ff_set_fixed_vector(p->fixed_vector, &fixed_sparse, 1.0,
1011
                            AMR_SUBFRAME_SIZE);
1012
1013
9088
        p->fixed_gain[4] =
1014
9088
            ff_amr_set_fixed_gain(fixed_gain_factor,
1015
9088
                       p->celpm_ctx.dot_productf(p->fixed_vector,
1016
9088
                                                               p->fixed_vector,
1017
                                                               AMR_SUBFRAME_SIZE) /
1018
                                  AMR_SUBFRAME_SIZE,
1019
9088
                       p->prediction_error,
1020
9088
                       energy_mean[p->cur_frame_mode], energy_pred_fac);
1021
1022
        // The excitation feedback is calculated without any processing such
1023
        // as fixed gain smoothing. This isn't mentioned in the specification.
1024
372608
        for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
1025
363520
            p->excitation[i] *= p->pitch_gain[4];
1026
9088
        ff_set_fixed_vector(p->excitation, &fixed_sparse, p->fixed_gain[4],
1027
                            AMR_SUBFRAME_SIZE);
1028
1029
        // In the ref decoder, excitation is stored with no fractional bits.
1030
        // This step prevents buzz in silent periods. The ref encoder can
1031
        // emit long sequences with pitch factor greater than one. This
1032
        // creates unwanted feedback if the excitation vector is nonzero.
1033
        // (e.g. test sequence T19_795.COD in 3GPP TS 26.074)
1034
372608
        for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
1035
363520
            p->excitation[i] = truncf(p->excitation[i]);
1036
1037
        // Smooth fixed gain.
1038
        // The specification is ambiguous, but in the reference source, the
1039
        // smoothed value is NOT fed back into later fixed gain smoothing.
1040
9088
        synth_fixed_gain = fixed_gain_smooth(p, p->lsf_q[subframe],
1041
9088
                                             p->lsf_avg, p->cur_frame_mode);
1042
1043
9088
        synth_fixed_vector = anti_sparseness(p, &fixed_sparse, p->fixed_vector,
1044
                                             synth_fixed_gain, spare_vector);
1045
1046
9088
        if (synthesis(p, p->lpc[subframe], synth_fixed_gain,
1047
                      synth_fixed_vector, &p->samples_in[LP_FILTER_ORDER], 0))
1048
            // overflow detected -> rerun synthesis scaling pitch vector down
1049
            // by a factor of 4, skipping pitch vector contribution emphasis
1050
            // and adaptive gain control
1051
            synthesis(p, p->lpc[subframe], synth_fixed_gain,
1052
                      synth_fixed_vector, &p->samples_in[LP_FILTER_ORDER], 1);
1053
1054
9088
        postfilter(p, p->lpc[subframe], buf_out + subframe * AMR_SUBFRAME_SIZE);
1055
1056
        // update buffers and history
1057
9088
        ff_clear_fixed_vector(p->fixed_vector, &fixed_sparse, AMR_SUBFRAME_SIZE);
1058
9088
        update_state(p);
1059
    }
1060
1061
2272
    p->acelpf_ctx.acelp_apply_order_2_transfer_function(buf_out,
1062
                                             buf_out, highpass_zeros,
1063
                                             highpass_poles,
1064
                                             highpass_gain * AMR_SAMPLE_SCALE,
1065
2272
                                             p->high_pass_mem, AMR_BLOCK_SIZE);
1066
1067
    /* Update averaged lsf vector (used for fixed gain smoothing).
1068
     *
1069
     * Note that lsf_avg should not incorporate the current frame's LSFs
1070
     * for fixed_gain_smooth.
1071
     * The specification has an incorrect formula: the reference decoder uses
1072
     * qbar(n-1) rather than qbar(n) in section 6.1(4) equation 71. */
1073
2272
    p->acelpv_ctx.weighted_vector_sumf(p->lsf_avg, p->lsf_avg, p->lsf_q[3],
1074
                            0.84, 0.16, LP_FILTER_ORDER);
1075
1076
2272
    *got_frame_ptr = 1;
1077
1078
    /* return the amount of bytes consumed if everything was OK */
1079
2272
    return frame_sizes_nb[p->cur_frame_mode] + 1; // +7 for rounding and +8 for TOC
1080
}
1081
1082
1083
AVCodec ff_amrnb_decoder = {
1084
    .name           = "amrnb",
1085
    .long_name      = NULL_IF_CONFIG_SMALL("AMR-NB (Adaptive Multi-Rate NarrowBand)"),
1086
    .type           = AVMEDIA_TYPE_AUDIO,
1087
    .id             = AV_CODEC_ID_AMR_NB,
1088
    .priv_data_size = sizeof(AMRContext),
1089
    .init           = amrnb_decode_init,
1090
    .decode         = amrnb_decode_frame,
1091
    .capabilities   = AV_CODEC_CAP_DR1,
1092
    .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT,
1093
                                                     AV_SAMPLE_FMT_NONE },
1094
};