1/*
2 * AMR narrowband decoder
3 * Copyright (c) 2006-2007 Robert Swain
4 * Copyright (c) 2009 Colin McQuillan
5 *
6 * 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
10 * License as published by the Free Software Foundation; either
11 * version 2.1 of the License, or (at your option) any later version.
12 *
13 * 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 "avcodec.h"
47#include "get_bits.h"
48#include "libavutil/common.h"
49#include "celp_math.h"
50#include "celp_filters.h"
51#include "acelp_filters.h"
52#include "acelp_vectors.h"
53#include "acelp_pitch_delay.h"
54#include "lsp.h"
55
56#include "amrnbdata.h"
57
58#define AMR_BLOCK_SIZE              160   ///< samples per frame
59#define AMR_SAMPLE_BOUND        32768.0   ///< threshold for synthesis overflow
60
61/**
62 * Scale from constructed speech to [-1,1]
63 *
64 * AMR is designed to produce 16-bit PCM samples (3GPP TS 26.090 4.2) but
65 * upscales by two (section 6.2.2).
66 *
67 * Fundamentally, this scale is determined by energy_mean through
68 * the fixed vector contribution to the excitation vector.
69 */
70#define AMR_SAMPLE_SCALE  (2.0 / 32768.0)
71
72/** Prediction factor for 12.2kbit/s mode */
73#define PRED_FAC_MODE_12k2             0.65
74
75#define LSF_R_FAC          (8000.0 / 32768.0) ///< LSF residual tables to Hertz
76#define MIN_LSF_SPACING    (50.0488 / 8000.0) ///< Ensures stability of LPC filter
77#define PITCH_LAG_MIN_MODE_12k2          18   ///< Lower bound on decoded lag search in 12.2kbit/s mode
78
79/** Initial energy in dB. Also used for bad frames (unimplemented). */
80#define MIN_ENERGY -14.0
81
82/** Maximum sharpening factor
83 *
84 * The specification says 0.8, which should be 13107, but the reference C code
85 * uses 13017 instead. (Amusingly the same applies to SHARP_MAX in g729dec.c.)
86 */
87#define SHARP_MAX 0.79449462890625
88
89/** Number of impulse response coefficients used for tilt factor */
90#define AMR_TILT_RESPONSE   22
91/** Tilt factor = 1st reflection coefficient * gamma_t */
92#define AMR_TILT_GAMMA_T   0.8
93/** Adaptive gain control factor used in post-filter */
94#define AMR_AGC_ALPHA      0.9
95
96typedef struct AMRContext {
97    AMRNBFrame                        frame; ///< decoded AMR parameters (lsf coefficients, codebook indexes, etc)
98    uint8_t             bad_frame_indicator; ///< bad frame ? 1 : 0
99    enum Mode                cur_frame_mode;
100
101    int16_t     prev_lsf_r[LP_FILTER_ORDER]; ///< residual LSF vector from previous subframe
102    double          lsp[4][LP_FILTER_ORDER]; ///< lsp vectors from current frame
103    double   prev_lsp_sub4[LP_FILTER_ORDER]; ///< lsp vector for the 4th subframe of the previous frame
104
105    float         lsf_q[4][LP_FILTER_ORDER]; ///< Interpolated LSF vector for fixed gain smoothing
106    float          lsf_avg[LP_FILTER_ORDER]; ///< vector of averaged lsf vector
107
108    float           lpc[4][LP_FILTER_ORDER]; ///< lpc coefficient vectors for 4 subframes
109
110    uint8_t                   pitch_lag_int; ///< integer part of pitch lag from current subframe
111
112    float excitation_buf[PITCH_DELAY_MAX + LP_FILTER_ORDER + 1 + AMR_SUBFRAME_SIZE]; ///< current excitation and all necessary excitation history
113    float                       *excitation; ///< pointer to the current excitation vector in excitation_buf
114
115    float   pitch_vector[AMR_SUBFRAME_SIZE]; ///< adaptive code book (pitch) vector
116    float   fixed_vector[AMR_SUBFRAME_SIZE]; ///< algebraic codebook (fixed) vector (must be kept zero between frames)
117
118    float               prediction_error[4]; ///< quantified prediction errors {20log10(^gamma_gc)} for previous four subframes
119    float                     pitch_gain[5]; ///< quantified pitch gains for the current and previous four subframes
120    float                     fixed_gain[5]; ///< quantified fixed gains for the current and previous four subframes
121
122    float                              beta; ///< previous pitch_gain, bounded by [0.0,SHARP_MAX]
123    uint8_t                      diff_count; ///< the number of subframes for which diff has been above 0.65
124    uint8_t                      hang_count; ///< the number of subframes since a hangover period started
125
126    float            prev_sparse_fixed_gain; ///< previous fixed gain; used by anti-sparseness processing to determine "onset"
127    uint8_t               prev_ir_filter_nr; ///< previous impulse response filter "impNr": 0 - strong, 1 - medium, 2 - none
128    uint8_t                 ir_filter_onset; ///< flag for impulse response filter strength
129
130    float                postfilter_mem[10]; ///< previous intermediate values in the formant filter
131    float                          tilt_mem; ///< previous input to tilt compensation filter
132    float                    postfilter_agc; ///< previous factor used for adaptive gain control
133    float                  high_pass_mem[2]; ///< previous intermediate values in the high-pass filter
134
135    float samples_in[LP_FILTER_ORDER + AMR_SUBFRAME_SIZE]; ///< floating point samples
136
137} AMRContext;
138
139/** Double version of ff_weighted_vector_sumf() */
140static void weighted_vector_sumd(double *out, const double *in_a,
141                                 const double *in_b, double weight_coeff_a,
142                                 double weight_coeff_b, int length)
143{
144    int i;
145
146    for (i = 0; i < length; i++)
147        out[i] = weight_coeff_a * in_a[i]
148               + weight_coeff_b * in_b[i];
149}
150
151static av_cold int amrnb_decode_init(AVCodecContext *avctx)
152{
153    AMRContext *p = avctx->priv_data;
154    int i;
155
156    avctx->sample_fmt = SAMPLE_FMT_FLT;
157
158    // p->excitation always points to the same position in p->excitation_buf
159    p->excitation = &p->excitation_buf[PITCH_DELAY_MAX + LP_FILTER_ORDER + 1];
160
161    for (i = 0; i < LP_FILTER_ORDER; i++) {
162        p->prev_lsp_sub4[i] =    lsp_sub4_init[i] * 1000 / (float)(1 << 15);
163        p->lsf_avg[i] = p->lsf_q[3][i] = lsp_avg_init[i] / (float)(1 << 15);
164    }
165
166    for (i = 0; i < 4; i++)
167        p->prediction_error[i] = MIN_ENERGY;
168
169    return 0;
170}
171
172
173/**
174 * Unpack an RFC4867 speech frame into the AMR frame mode and parameters.
175 *
176 * The order of speech bits is specified by 3GPP TS 26.101.
177 *
178 * @param p the context
179 * @param buf               pointer to the input buffer
180 * @param buf_size          size of the input buffer
181 *
182 * @return the frame mode
183 */
184static enum Mode unpack_bitstream(AMRContext *p, const uint8_t *buf,
185                                  int buf_size)
186{
187    GetBitContext gb;
188    enum Mode mode;
189
190    init_get_bits(&gb, buf, buf_size * 8);
191
192    // Decode the first octet.
193    skip_bits(&gb, 1);                        // padding bit
194    mode = get_bits(&gb, 4);                  // frame type
195    p->bad_frame_indicator = !get_bits1(&gb); // quality bit
196    skip_bits(&gb, 2);                        // two padding bits
197
198    if (mode < MODE_DTX) {
199        uint16_t *data = (uint16_t *)&p->frame;
200        const uint8_t *order = amr_unpacking_bitmaps_per_mode[mode];
201        int field_size;
202
203        memset(&p->frame, 0, sizeof(AMRNBFrame));
204        buf++;
205        while ((field_size = *order++)) {
206            int field = 0;
207            int field_offset = *order++;
208            while (field_size--) {
209               int bit = *order++;
210               field <<= 1;
211               field |= buf[bit >> 3] >> (bit & 7) & 1;
212            }
213            data[field_offset] = field;
214        }
215    }
216
217    return mode;
218}
219
220
221/// @defgroup amr_lpc_decoding AMR pitch LPC coefficient decoding functions
222/// @{
223
224/**
225 * Convert an lsf vector into an lsp vector.
226 *
227 * @param lsf               input lsf vector
228 * @param lsp               output lsp vector
229 */
230static void lsf2lsp(const float *lsf, double *lsp)
231{
232    int i;
233
234    for (i = 0; i < LP_FILTER_ORDER; i++)
235        lsp[i] = cos(2.0 * M_PI * lsf[i]);
236}
237
238/**
239 * Interpolate the LSF vector (used for fixed gain smoothing).
240 * The interpolation is done over all four subframes even in MODE_12k2.
241 *
242 * @param[in,out] lsf_q     LSFs in [0,1] for each subframe
243 * @param[in]     lsf_new   New LSFs in [0,1] for subframe 4
244 */
245static void interpolate_lsf(float lsf_q[4][LP_FILTER_ORDER], float *lsf_new)
246{
247    int i;
248
249    for (i = 0; i < 4; i++)
250        ff_weighted_vector_sumf(lsf_q[i], lsf_q[3], lsf_new,
251                                0.25 * (3 - i), 0.25 * (i + 1),
252                                LP_FILTER_ORDER);
253}
254
255/**
256 * Decode a set of 5 split-matrix quantized lsf indexes into an lsp vector.
257 *
258 * @param p the context
259 * @param lsp output LSP vector
260 * @param lsf_no_r LSF vector without the residual vector added
261 * @param lsf_quantizer pointers to LSF dictionary tables
262 * @param quantizer_offset offset in tables
263 * @param sign for the 3 dictionary table
264 * @param update store data for computing the next frame's LSFs
265 */
266static void lsf2lsp_for_mode12k2(AMRContext *p, double lsp[LP_FILTER_ORDER],
267                                 const float lsf_no_r[LP_FILTER_ORDER],
268                                 const int16_t *lsf_quantizer[5],
269                                 const int quantizer_offset,
270                                 const int sign, const int update)
271{
272    int16_t lsf_r[LP_FILTER_ORDER]; // residual LSF vector
273    float lsf_q[LP_FILTER_ORDER]; // quantified LSF vector
274    int i;
275
276    for (i = 0; i < LP_FILTER_ORDER >> 1; i++)
277        memcpy(&lsf_r[i << 1], &lsf_quantizer[i][quantizer_offset],
278               2 * sizeof(*lsf_r));
279
280    if (sign) {
281        lsf_r[4] *= -1;
282        lsf_r[5] *= -1;
283    }
284
285    if (update)
286        memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(float));
287
288    for (i = 0; i < LP_FILTER_ORDER; i++)
289        lsf_q[i] = lsf_r[i] * (LSF_R_FAC / 8000.0) + lsf_no_r[i] * (1.0 / 8000.0);
290
291    ff_set_min_dist_lsf(lsf_q, MIN_LSF_SPACING, LP_FILTER_ORDER);
292
293    if (update)
294        interpolate_lsf(p->lsf_q, lsf_q);
295
296    lsf2lsp(lsf_q, lsp);
297}
298
299/**
300 * Decode a set of 5 split-matrix quantized lsf indexes into 2 lsp vectors.
301 *
302 * @param p                 pointer to the AMRContext
303 */
304static void lsf2lsp_5(AMRContext *p)
305{
306    const uint16_t *lsf_param = p->frame.lsf;
307    float lsf_no_r[LP_FILTER_ORDER]; // LSFs without the residual vector
308    const int16_t *lsf_quantizer[5];
309    int i;
310
311    lsf_quantizer[0] = lsf_5_1[lsf_param[0]];
312    lsf_quantizer[1] = lsf_5_2[lsf_param[1]];
313    lsf_quantizer[2] = lsf_5_3[lsf_param[2] >> 1];
314    lsf_quantizer[3] = lsf_5_4[lsf_param[3]];
315    lsf_quantizer[4] = lsf_5_5[lsf_param[4]];
316
317    for (i = 0; i < LP_FILTER_ORDER; i++)
318        lsf_no_r[i] = p->prev_lsf_r[i] * LSF_R_FAC * PRED_FAC_MODE_12k2 + lsf_5_mean[i];
319
320    lsf2lsp_for_mode12k2(p, p->lsp[1], lsf_no_r, lsf_quantizer, 0, lsf_param[2] & 1, 0);
321    lsf2lsp_for_mode12k2(p, p->lsp[3], lsf_no_r, lsf_quantizer, 2, lsf_param[2] & 1, 1);
322
323    // interpolate LSP vectors at subframes 1 and 3
324    weighted_vector_sumd(p->lsp[0], p->prev_lsp_sub4, p->lsp[1], 0.5, 0.5, LP_FILTER_ORDER);
325    weighted_vector_sumd(p->lsp[2], p->lsp[1]       , p->lsp[3], 0.5, 0.5, LP_FILTER_ORDER);
326}
327
328/**
329 * Decode a set of 3 split-matrix quantized lsf indexes into an lsp vector.
330 *
331 * @param p                 pointer to the AMRContext
332 */
333static void lsf2lsp_3(AMRContext *p)
334{
335    const uint16_t *lsf_param = p->frame.lsf;
336    int16_t lsf_r[LP_FILTER_ORDER]; // residual LSF vector
337    float lsf_q[LP_FILTER_ORDER]; // quantified LSF vector
338    const int16_t *lsf_quantizer;
339    int i, j;
340
341    lsf_quantizer = (p->cur_frame_mode == MODE_7k95 ? lsf_3_1_MODE_7k95 : lsf_3_1)[lsf_param[0]];
342    memcpy(lsf_r, lsf_quantizer, 3 * sizeof(*lsf_r));
343
344    lsf_quantizer = lsf_3_2[lsf_param[1] << (p->cur_frame_mode <= MODE_5k15)];
345    memcpy(lsf_r + 3, lsf_quantizer, 3 * sizeof(*lsf_r));
346
347    lsf_quantizer = (p->cur_frame_mode <= MODE_5k15 ? lsf_3_3_MODE_5k15 : lsf_3_3)[lsf_param[2]];
348    memcpy(lsf_r + 6, lsf_quantizer, 4 * sizeof(*lsf_r));
349
350    // calculate mean-removed LSF vector and add mean
351    for (i = 0; i < LP_FILTER_ORDER; i++)
352        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);
353
354    ff_set_min_dist_lsf(lsf_q, MIN_LSF_SPACING, LP_FILTER_ORDER);
355
356    // store data for computing the next frame's LSFs
357    interpolate_lsf(p->lsf_q, lsf_q);
358    memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(*lsf_r));
359
360    lsf2lsp(lsf_q, p->lsp[3]);
361
362    // interpolate LSP vectors at subframes 1, 2 and 3
363    for (i = 1; i <= 3; i++)
364        for(j = 0; j < LP_FILTER_ORDER; j++)
365            p->lsp[i-1][j] = p->prev_lsp_sub4[j] +
366                (p->lsp[3][j] - p->prev_lsp_sub4[j]) * 0.25 * i;
367}
368
369/// @}
370
371
372/// @defgroup amr_pitch_vector_decoding AMR pitch vector decoding functions
373/// @{
374
375/**
376 * Like ff_decode_pitch_lag(), but with 1/6 resolution
377 */
378static void decode_pitch_lag_1_6(int *lag_int, int *lag_frac, int pitch_index,
379                                 const int prev_lag_int, const int subframe)
380{
381    if (subframe == 0 || subframe == 2) {
382        if (pitch_index < 463) {
383            *lag_int  = (pitch_index + 107) * 10923 >> 16;
384            *lag_frac = pitch_index - *lag_int * 6 + 105;
385        } else {
386            *lag_int  = pitch_index - 368;
387            *lag_frac = 0;
388        }
389    } else {
390        *lag_int  = ((pitch_index + 5) * 10923 >> 16) - 1;
391        *lag_frac = pitch_index - *lag_int * 6 - 3;
392        *lag_int += av_clip(prev_lag_int - 5, PITCH_LAG_MIN_MODE_12k2,
393                            PITCH_DELAY_MAX - 9);
394    }
395}
396
397static void decode_pitch_vector(AMRContext *p,
398                                const AMRNBSubframe *amr_subframe,
399                                const int subframe)
400{
401    int pitch_lag_int, pitch_lag_frac;
402    enum Mode mode = p->cur_frame_mode;
403
404    if (p->cur_frame_mode == MODE_12k2) {
405        decode_pitch_lag_1_6(&pitch_lag_int, &pitch_lag_frac,
406                             amr_subframe->p_lag, p->pitch_lag_int,
407                             subframe);
408    } else
409        ff_decode_pitch_lag(&pitch_lag_int, &pitch_lag_frac,
410                            amr_subframe->p_lag,
411                            p->pitch_lag_int, subframe,
412                            mode != MODE_4k75 && mode != MODE_5k15,
413                            mode <= MODE_6k7 ? 4 : (mode == MODE_7k95 ? 5 : 6));
414
415    p->pitch_lag_int = pitch_lag_int; // store previous lag in a uint8_t
416
417    pitch_lag_frac <<= (p->cur_frame_mode != MODE_12k2);
418
419    pitch_lag_int += pitch_lag_frac > 0;
420
421    /* Calculate the pitch vector by interpolating the past excitation at the
422       pitch lag using a b60 hamming windowed sinc function.   */
423    ff_acelp_interpolatef(p->excitation, p->excitation + 1 - pitch_lag_int,
424                          ff_b60_sinc, 6,
425                          pitch_lag_frac + 6 - 6*(pitch_lag_frac > 0),
426                          10, AMR_SUBFRAME_SIZE);
427
428    memcpy(p->pitch_vector, p->excitation, AMR_SUBFRAME_SIZE * sizeof(float));
429}
430
431/// @}
432
433
434/// @defgroup amr_algebraic_code_book AMR algebraic code book (fixed) vector decoding functions
435/// @{
436
437/**
438 * Decode a 10-bit algebraic codebook index from a 10.2 kbit/s frame.
439 */
440static void decode_10bit_pulse(int code, int pulse_position[8],
441                               int i1, int i2, int i3)
442{
443    // coded using 7+3 bits with the 3 LSBs being, individually, the LSB of 1 of
444    // the 3 pulses and the upper 7 bits being coded in base 5
445    const uint8_t *positions = base_five_table[code >> 3];
446    pulse_position[i1] = (positions[2] << 1) + ( code       & 1);
447    pulse_position[i2] = (positions[1] << 1) + ((code >> 1) & 1);
448    pulse_position[i3] = (positions[0] << 1) + ((code >> 2) & 1);
449}
450
451/**
452 * Decode the algebraic codebook index to pulse positions and signs and
453 * construct the algebraic codebook vector for MODE_10k2.
454 *
455 * @param fixed_index          positions of the eight pulses
456 * @param fixed_sparse         pointer to the algebraic codebook vector
457 */
458static void decode_8_pulses_31bits(const int16_t *fixed_index,
459                                   AMRFixed *fixed_sparse)
460{
461    int pulse_position[8];
462    int i, temp;
463
464    decode_10bit_pulse(fixed_index[4], pulse_position, 0, 4, 1);
465    decode_10bit_pulse(fixed_index[5], pulse_position, 2, 6, 5);
466
467    // coded using 5+2 bits with the 2 LSBs being, individually, the LSB of 1 of
468    // the 2 pulses and the upper 5 bits being coded in base 5
469    temp = ((fixed_index[6] >> 2) * 25 + 12) >> 5;
470    pulse_position[3] = temp % 5;
471    pulse_position[7] = temp / 5;
472    if (pulse_position[7] & 1)
473        pulse_position[3] = 4 - pulse_position[3];
474    pulse_position[3] = (pulse_position[3] << 1) + ( fixed_index[6]       & 1);
475    pulse_position[7] = (pulse_position[7] << 1) + ((fixed_index[6] >> 1) & 1);
476
477    fixed_sparse->n = 8;
478    for (i = 0; i < 4; i++) {
479        const int pos1   = (pulse_position[i]     << 2) + i;
480        const int pos2   = (pulse_position[i + 4] << 2) + i;
481        const float sign = fixed_index[i] ? -1.0 : 1.0;
482        fixed_sparse->x[i    ] = pos1;
483        fixed_sparse->x[i + 4] = pos2;
484        fixed_sparse->y[i    ] = sign;
485        fixed_sparse->y[i + 4] = pos2 < pos1 ? -sign : sign;
486    }
487}
488
489/**
490 * Decode the algebraic codebook index to pulse positions and signs,
491 * then construct the algebraic codebook vector.
492 *
493 *                              nb of pulses | bits encoding pulses
494 * For MODE_4k75 or MODE_5k15,             2 | 1-3, 4-6, 7
495 *                  MODE_5k9,              2 | 1,   2-4, 5-6, 7-9
496 *                  MODE_6k7,              3 | 1-3, 4,   5-7, 8,  9-11
497 *      MODE_7k4 or MODE_7k95,             4 | 1-3, 4-6, 7-9, 10, 11-13
498 *
499 * @param fixed_sparse pointer to the algebraic codebook vector
500 * @param pulses       algebraic codebook indexes
501 * @param mode         mode of the current frame
502 * @param subframe     current subframe number
503 */
504static void decode_fixed_sparse(AMRFixed *fixed_sparse, const uint16_t *pulses,
505                                const enum Mode mode, const int subframe)
506{
507    assert(MODE_4k75 <= mode && mode <= MODE_12k2);
508
509    if (mode == MODE_12k2) {
510        ff_decode_10_pulses_35bits(pulses, fixed_sparse, gray_decode, 5, 3);
511    } else if (mode == MODE_10k2) {
512        decode_8_pulses_31bits(pulses, fixed_sparse);
513    } else {
514        int *pulse_position = fixed_sparse->x;
515        int i, pulse_subset;
516        const int fixed_index = pulses[0];
517
518        if (mode <= MODE_5k15) {
519            pulse_subset      = ((fixed_index >> 3) & 8)     + (subframe << 1);
520            pulse_position[0] = ( fixed_index       & 7) * 5 + track_position[pulse_subset];
521            pulse_position[1] = ((fixed_index >> 3) & 7) * 5 + track_position[pulse_subset + 1];
522            fixed_sparse->n = 2;
523        } else if (mode == MODE_5k9) {
524            pulse_subset      = ((fixed_index & 1) << 1) + 1;
525            pulse_position[0] = ((fixed_index >> 1) & 7) * 5 + pulse_subset;
526            pulse_subset      = (fixed_index  >> 4) & 3;
527            pulse_position[1] = ((fixed_index >> 6) & 7) * 5 + pulse_subset + (pulse_subset == 3 ? 1 : 0);
528            fixed_sparse->n = pulse_position[0] == pulse_position[1] ? 1 : 2;
529        } else if (mode == MODE_6k7) {
530            pulse_position[0] = (fixed_index        & 7) * 5;
531            pulse_subset      = (fixed_index  >> 2) & 2;
532            pulse_position[1] = ((fixed_index >> 4) & 7) * 5 + pulse_subset + 1;
533            pulse_subset      = (fixed_index  >> 6) & 2;
534            pulse_position[2] = ((fixed_index >> 8) & 7) * 5 + pulse_subset + 2;
535            fixed_sparse->n = 3;
536        } else { // mode <= MODE_7k95
537            pulse_position[0] = gray_decode[ fixed_index        & 7];
538            pulse_position[1] = gray_decode[(fixed_index >> 3)  & 7] + 1;
539            pulse_position[2] = gray_decode[(fixed_index >> 6)  & 7] + 2;
540            pulse_subset      = (fixed_index >> 9) & 1;
541            pulse_position[3] = gray_decode[(fixed_index >> 10) & 7] + pulse_subset + 3;
542            fixed_sparse->n = 4;
543        }
544        for (i = 0; i < fixed_sparse->n; i++)
545            fixed_sparse->y[i] = (pulses[1] >> i) & 1 ? 1.0 : -1.0;
546    }
547}
548
549/**
550 * Apply pitch lag to obtain the sharpened fixed vector (section 6.1.2)
551 *
552 * @param p the context
553 * @param subframe unpacked amr subframe
554 * @param mode mode of the current frame
555 * @param fixed_sparse sparse respresentation of the fixed vector
556 */
557static void pitch_sharpening(AMRContext *p, int subframe, enum Mode mode,
558                             AMRFixed *fixed_sparse)
559{
560    // The spec suggests the current pitch gain is always used, but in other
561    // modes the pitch and codebook gains are joinly quantized (sec 5.8.2)
562    // so the codebook gain cannot depend on the quantized pitch gain.
563    if (mode == MODE_12k2)
564        p->beta = FFMIN(p->pitch_gain[4], 1.0);
565
566    fixed_sparse->pitch_lag  = p->pitch_lag_int;
567    fixed_sparse->pitch_fac  = p->beta;
568
569    // Save pitch sharpening factor for the next subframe
570    // MODE_4k75 only updates on the 2nd and 4th subframes - this follows from
571    // the fact that the gains for two subframes are jointly quantized.
572    if (mode != MODE_4k75 || subframe & 1)
573        p->beta = av_clipf(p->pitch_gain[4], 0.0, SHARP_MAX);
574}
575/// @}
576
577
578/// @defgroup amr_gain_decoding AMR gain decoding functions
579/// @{
580
581/**
582 * fixed gain smoothing
583 * Note that where the spec specifies the "spectrum in the q domain"
584 * in section 6.1.4, in fact frequencies should be used.
585 *
586 * @param p the context
587 * @param lsf LSFs for the current subframe, in the range [0,1]
588 * @param lsf_avg averaged LSFs
589 * @param mode mode of the current frame
590 *
591 * @return fixed gain smoothed
592 */
593static float fixed_gain_smooth(AMRContext *p , const float *lsf,
594                               const float *lsf_avg, const enum Mode mode)
595{
596    float diff = 0.0;
597    int i;
598
599    for (i = 0; i < LP_FILTER_ORDER; i++)
600        diff += fabs(lsf_avg[i] - lsf[i]) / lsf_avg[i];
601
602    // If diff is large for ten subframes, disable smoothing for a 40-subframe
603    // hangover period.
604    p->diff_count++;
605    if (diff <= 0.65)
606        p->diff_count = 0;
607
608    if (p->diff_count > 10) {
609        p->hang_count = 0;
610        p->diff_count--; // don't let diff_count overflow
611    }
612
613    if (p->hang_count < 40) {
614        p->hang_count++;
615    } else if (mode < MODE_7k4 || mode == MODE_10k2) {
616        const float smoothing_factor = av_clipf(4.0 * diff - 1.6, 0.0, 1.0);
617        const float fixed_gain_mean = (p->fixed_gain[0] + p->fixed_gain[1] +
618                                       p->fixed_gain[2] + p->fixed_gain[3] +
619                                       p->fixed_gain[4]) * 0.2;
620        return smoothing_factor * p->fixed_gain[4] +
621               (1.0 - smoothing_factor) * fixed_gain_mean;
622    }
623    return p->fixed_gain[4];
624}
625
626/**
627 * Decode pitch gain and fixed gain factor (part of section 6.1.3).
628 *
629 * @param p the context
630 * @param amr_subframe unpacked amr subframe
631 * @param mode mode of the current frame
632 * @param subframe current subframe number
633 * @param fixed_gain_factor decoded gain correction factor
634 */
635static void decode_gains(AMRContext *p, const AMRNBSubframe *amr_subframe,
636                         const enum Mode mode, const int subframe,
637                         float *fixed_gain_factor)
638{
639    if (mode == MODE_12k2 || mode == MODE_7k95) {
640        p->pitch_gain[4]   = qua_gain_pit [amr_subframe->p_gain    ]
641            * (1.0 / 16384.0);
642        *fixed_gain_factor = qua_gain_code[amr_subframe->fixed_gain]
643            * (1.0 /  2048.0);
644    } else {
645        const uint16_t *gains;
646
647        if (mode >= MODE_6k7) {
648            gains = gains_high[amr_subframe->p_gain];
649        } else if (mode >= MODE_5k15) {
650            gains = gains_low [amr_subframe->p_gain];
651        } else {
652            // gain index is only coded in subframes 0,2 for MODE_4k75
653            gains = gains_MODE_4k75[(p->frame.subframe[subframe & 2].p_gain << 1) + (subframe & 1)];
654        }
655
656        p->pitch_gain[4]   = gains[0] * (1.0 / 16384.0);
657        *fixed_gain_factor = gains[1] * (1.0 /  4096.0);
658    }
659}
660
661/// @}
662
663
664/// @defgroup amr_pre_processing AMR pre-processing functions
665/// @{
666
667/**
668 * Circularly convolve a sparse fixed vector with a phase dispersion impulse
669 * response filter (D.6.2 of G.729 and 6.1.5 of AMR).
670 *
671 * @param out vector with filter applied
672 * @param in source vector
673 * @param filter phase filter coefficients
674 *
675 *  out[n] = sum(i,0,len-1){ in[i] * filter[(len + n - i)%len] }
676 */
677static void apply_ir_filter(float *out, const AMRFixed *in,
678                            const float *filter)
679{
680    float filter1[AMR_SUBFRAME_SIZE],     //!< filters at pitch lag*1 and *2
681          filter2[AMR_SUBFRAME_SIZE];
682    int   lag = in->pitch_lag;
683    float fac = in->pitch_fac;
684    int i;
685
686    if (lag < AMR_SUBFRAME_SIZE) {
687        ff_celp_circ_addf(filter1, filter, filter, lag, fac,
688                          AMR_SUBFRAME_SIZE);
689
690        if (lag < AMR_SUBFRAME_SIZE >> 1)
691            ff_celp_circ_addf(filter2, filter, filter1, lag, fac,
692                              AMR_SUBFRAME_SIZE);
693    }
694
695    memset(out, 0, sizeof(float) * AMR_SUBFRAME_SIZE);
696    for (i = 0; i < in->n; i++) {
697        int   x = in->x[i];
698        float y = in->y[i];
699        const float *filterp;
700
701        if (x >= AMR_SUBFRAME_SIZE - lag) {
702            filterp = filter;
703        } else if (x >= AMR_SUBFRAME_SIZE - (lag << 1)) {
704            filterp = filter1;
705        } else
706            filterp = filter2;
707
708        ff_celp_circ_addf(out, out, filterp, x, y, AMR_SUBFRAME_SIZE);
709    }
710}
711
712/**
713 * Reduce fixed vector sparseness by smoothing with one of three IR filters.
714 * Also know as "adaptive phase dispersion".
715 *
716 * This implements 3GPP TS 26.090 section 6.1(5).
717 *
718 * @param p the context
719 * @param fixed_sparse algebraic codebook vector
720 * @param fixed_vector unfiltered fixed vector
721 * @param fixed_gain smoothed gain
722 * @param out space for modified vector if necessary
723 */
724static const float *anti_sparseness(AMRContext *p, AMRFixed *fixed_sparse,
725                                    const float *fixed_vector,
726                                    float fixed_gain, float *out)
727{
728    int ir_filter_nr;
729
730    if (p->pitch_gain[4] < 0.6) {
731        ir_filter_nr = 0;      // strong filtering
732    } else if (p->pitch_gain[4] < 0.9) {
733        ir_filter_nr = 1;      // medium filtering
734    } else
735        ir_filter_nr = 2;      // no filtering
736
737    // detect 'onset'
738    if (fixed_gain > 2.0 * p->prev_sparse_fixed_gain) {
739        p->ir_filter_onset = 2;
740    } else if (p->ir_filter_onset)
741        p->ir_filter_onset--;
742
743    if (!p->ir_filter_onset) {
744        int i, count = 0;
745
746        for (i = 0; i < 5; i++)
747            if (p->pitch_gain[i] < 0.6)
748                count++;
749        if (count > 2)
750            ir_filter_nr = 0;
751
752        if (ir_filter_nr > p->prev_ir_filter_nr + 1)
753            ir_filter_nr--;
754    } else if (ir_filter_nr < 2)
755        ir_filter_nr++;
756
757    // Disable filtering for very low level of fixed_gain.
758    // Note this step is not specified in the technical description but is in
759    // the reference source in the function Ph_disp.
760    if (fixed_gain < 5.0)
761        ir_filter_nr = 2;
762
763    if (p->cur_frame_mode != MODE_7k4 && p->cur_frame_mode < MODE_10k2
764         && ir_filter_nr < 2) {
765        apply_ir_filter(out, fixed_sparse,
766                        (p->cur_frame_mode == MODE_7k95 ?
767                             ir_filters_lookup_MODE_7k95 :
768                             ir_filters_lookup)[ir_filter_nr]);
769        fixed_vector = out;
770    }
771
772    // update ir filter strength history
773    p->prev_ir_filter_nr       = ir_filter_nr;
774    p->prev_sparse_fixed_gain  = fixed_gain;
775
776    return fixed_vector;
777}
778
779/// @}
780
781
782/// @defgroup amr_synthesis AMR synthesis functions
783/// @{
784
785/**
786 * Conduct 10th order linear predictive coding synthesis.
787 *
788 * @param p             pointer to the AMRContext
789 * @param lpc           pointer to the LPC coefficients
790 * @param fixed_gain    fixed codebook gain for synthesis
791 * @param fixed_vector  algebraic codebook vector
792 * @param samples       pointer to the output speech samples
793 * @param overflow      16-bit overflow flag
794 */
795static int synthesis(AMRContext *p, float *lpc,
796                     float fixed_gain, const float *fixed_vector,
797                     float *samples, uint8_t overflow)
798{
799    int i;
800    float excitation[AMR_SUBFRAME_SIZE];
801
802    // if an overflow has been detected, the pitch vector is scaled down by a
803    // factor of 4
804    if (overflow)
805        for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
806            p->pitch_vector[i] *= 0.25;
807
808    ff_weighted_vector_sumf(excitation, p->pitch_vector, fixed_vector,
809                            p->pitch_gain[4], fixed_gain, AMR_SUBFRAME_SIZE);
810
811    // emphasize pitch vector contribution
812    if (p->pitch_gain[4] > 0.5 && !overflow) {
813        float energy = ff_dot_productf(excitation, excitation,
814                                       AMR_SUBFRAME_SIZE);
815        float pitch_factor =
816            p->pitch_gain[4] *
817            (p->cur_frame_mode == MODE_12k2 ?
818                0.25 * FFMIN(p->pitch_gain[4], 1.0) :
819                0.5  * FFMIN(p->pitch_gain[4], SHARP_MAX));
820
821        for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
822            excitation[i] += pitch_factor * p->pitch_vector[i];
823
824        ff_scale_vector_to_given_sum_of_squares(excitation, excitation, energy,
825                                                AMR_SUBFRAME_SIZE);
826    }
827
828    ff_celp_lp_synthesis_filterf(samples, lpc, excitation, AMR_SUBFRAME_SIZE,
829                                 LP_FILTER_ORDER);
830
831    // detect overflow
832    for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
833        if (fabsf(samples[i]) > AMR_SAMPLE_BOUND) {
834            return 1;
835        }
836
837    return 0;
838}
839
840/// @}
841
842
843/// @defgroup amr_update AMR update functions
844/// @{
845
846/**
847 * Update buffers and history at the end of decoding a subframe.
848 *
849 * @param p             pointer to the AMRContext
850 */
851static void update_state(AMRContext *p)
852{
853    memcpy(p->prev_lsp_sub4, p->lsp[3], LP_FILTER_ORDER * sizeof(p->lsp[3][0]));
854
855    memmove(&p->excitation_buf[0], &p->excitation_buf[AMR_SUBFRAME_SIZE],
856            (PITCH_DELAY_MAX + LP_FILTER_ORDER + 1) * sizeof(float));
857
858    memmove(&p->pitch_gain[0], &p->pitch_gain[1], 4 * sizeof(float));
859    memmove(&p->fixed_gain[0], &p->fixed_gain[1], 4 * sizeof(float));
860
861    memmove(&p->samples_in[0], &p->samples_in[AMR_SUBFRAME_SIZE],
862            LP_FILTER_ORDER * sizeof(float));
863}
864
865/// @}
866
867
868/// @defgroup amr_postproc AMR Post processing functions
869/// @{
870
871/**
872 * Get the tilt factor of a formant filter from its transfer function
873 *
874 * @param lpc_n LP_FILTER_ORDER coefficients of the numerator
875 * @param lpc_d LP_FILTER_ORDER coefficients of the denominator
876 */
877static float tilt_factor(float *lpc_n, float *lpc_d)
878{
879    float rh0, rh1; // autocorrelation at lag 0 and 1
880
881    // LP_FILTER_ORDER prior zeros are needed for ff_celp_lp_synthesis_filterf
882    float impulse_buffer[LP_FILTER_ORDER + AMR_TILT_RESPONSE] = { 0 };
883    float *hf = impulse_buffer + LP_FILTER_ORDER; // start of impulse response
884
885    hf[0] = 1.0;
886    memcpy(hf + 1, lpc_n, sizeof(float) * LP_FILTER_ORDER);
887    ff_celp_lp_synthesis_filterf(hf, lpc_d, hf, AMR_TILT_RESPONSE,
888                                 LP_FILTER_ORDER);
889
890    rh0 = ff_dot_productf(hf, hf,     AMR_TILT_RESPONSE);
891    rh1 = ff_dot_productf(hf, hf + 1, AMR_TILT_RESPONSE - 1);
892
893    // The spec only specifies this check for 12.2 and 10.2 kbit/s
894    // modes. But in the ref source the tilt is always non-negative.
895    return rh1 >= 0.0 ? rh1 / rh0 * AMR_TILT_GAMMA_T : 0.0;
896}
897
898/**
899 * Perform adaptive post-filtering to enhance the quality of the speech.
900 * See section 6.2.1.
901 *
902 * @param p             pointer to the AMRContext
903 * @param lpc           interpolated LP coefficients for this subframe
904 * @param buf_out       output of the filter
905 */
906static void postfilter(AMRContext *p, float *lpc, float *buf_out)
907{
908    int i;
909    float *samples          = p->samples_in + LP_FILTER_ORDER; // Start of input
910
911    float speech_gain       = ff_dot_productf(samples, samples,
912                                              AMR_SUBFRAME_SIZE);
913
914    float pole_out[AMR_SUBFRAME_SIZE + LP_FILTER_ORDER];  // Output of pole filter
915    const float *gamma_n, *gamma_d;                       // Formant filter factor table
916    float lpc_n[LP_FILTER_ORDER], lpc_d[LP_FILTER_ORDER]; // Transfer function coefficients
917
918    if (p->cur_frame_mode == MODE_12k2 || p->cur_frame_mode == MODE_10k2) {
919        gamma_n = ff_pow_0_7;
920        gamma_d = ff_pow_0_75;
921    } else {
922        gamma_n = ff_pow_0_55;
923        gamma_d = ff_pow_0_7;
924    }
925
926    for (i = 0; i < LP_FILTER_ORDER; i++) {
927         lpc_n[i] = lpc[i] * gamma_n[i];
928         lpc_d[i] = lpc[i] * gamma_d[i];
929    }
930
931    memcpy(pole_out, p->postfilter_mem, sizeof(float) * LP_FILTER_ORDER);
932    ff_celp_lp_synthesis_filterf(pole_out + LP_FILTER_ORDER, lpc_d, samples,
933                                 AMR_SUBFRAME_SIZE, LP_FILTER_ORDER);
934    memcpy(p->postfilter_mem, pole_out + AMR_SUBFRAME_SIZE,
935           sizeof(float) * LP_FILTER_ORDER);
936
937    ff_celp_lp_zero_synthesis_filterf(buf_out, lpc_n,
938                                      pole_out + LP_FILTER_ORDER,
939                                      AMR_SUBFRAME_SIZE, LP_FILTER_ORDER);
940
941    ff_tilt_compensation(&p->tilt_mem, tilt_factor(lpc_n, lpc_d), buf_out,
942                         AMR_SUBFRAME_SIZE);
943
944    ff_adaptive_gain_control(buf_out, buf_out, speech_gain, AMR_SUBFRAME_SIZE,
945                             AMR_AGC_ALPHA, &p->postfilter_agc);
946}
947
948/// @}
949
950static int amrnb_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
951                              AVPacket *avpkt)
952{
953
954    AMRContext *p = avctx->priv_data;        // pointer to private data
955    const uint8_t *buf = avpkt->data;
956    int buf_size       = avpkt->size;
957    float *buf_out = data;                   // pointer to the output data buffer
958    int i, subframe;
959    float fixed_gain_factor;
960    AMRFixed fixed_sparse = {0};             // fixed vector up to anti-sparseness processing
961    float spare_vector[AMR_SUBFRAME_SIZE];   // extra stack space to hold result from anti-sparseness processing
962    float synth_fixed_gain;                  // the fixed gain that synthesis should use
963    const float *synth_fixed_vector;         // pointer to the fixed vector that synthesis should use
964
965    p->cur_frame_mode = unpack_bitstream(p, buf, buf_size);
966    if (p->cur_frame_mode == MODE_DTX) {
967        av_log_missing_feature(avctx, "dtx mode", 1);
968        return -1;
969    }
970
971    if (p->cur_frame_mode == MODE_12k2) {
972        lsf2lsp_5(p);
973    } else
974        lsf2lsp_3(p);
975
976    for (i = 0; i < 4; i++)
977        ff_acelp_lspd2lpc(p->lsp[i], p->lpc[i], 5);
978
979    for (subframe = 0; subframe < 4; subframe++) {
980        const AMRNBSubframe *amr_subframe = &p->frame.subframe[subframe];
981
982        decode_pitch_vector(p, amr_subframe, subframe);
983
984        decode_fixed_sparse(&fixed_sparse, amr_subframe->pulses,
985                            p->cur_frame_mode, subframe);
986
987        // The fixed gain (section 6.1.3) depends on the fixed vector
988        // (section 6.1.2), but the fixed vector calculation uses
989        // pitch sharpening based on the on the pitch gain (section 6.1.3).
990        // So the correct order is: pitch gain, pitch sharpening, fixed gain.
991        decode_gains(p, amr_subframe, p->cur_frame_mode, subframe,
992                     &fixed_gain_factor);
993
994        pitch_sharpening(p, subframe, p->cur_frame_mode, &fixed_sparse);
995
996        ff_set_fixed_vector(p->fixed_vector, &fixed_sparse, 1.0,
997                            AMR_SUBFRAME_SIZE);
998
999        p->fixed_gain[4] =
1000            ff_amr_set_fixed_gain(fixed_gain_factor,
1001                       ff_dot_productf(p->fixed_vector, p->fixed_vector,
1002                                       AMR_SUBFRAME_SIZE)/AMR_SUBFRAME_SIZE,
1003                       p->prediction_error,
1004                       energy_mean[p->cur_frame_mode], energy_pred_fac);
1005
1006        // The excitation feedback is calculated without any processing such
1007        // as fixed gain smoothing. This isn't mentioned in the specification.
1008        for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
1009            p->excitation[i] *= p->pitch_gain[4];
1010        ff_set_fixed_vector(p->excitation, &fixed_sparse, p->fixed_gain[4],
1011                            AMR_SUBFRAME_SIZE);
1012
1013        // In the ref decoder, excitation is stored with no fractional bits.
1014        // This step prevents buzz in silent periods. The ref encoder can
1015        // emit long sequences with pitch factor greater than one. This
1016        // creates unwanted feedback if the excitation vector is nonzero.
1017        // (e.g. test sequence T19_795.COD in 3GPP TS 26.074)
1018        for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
1019            p->excitation[i] = truncf(p->excitation[i]);
1020
1021        // Smooth fixed gain.
1022        // The specification is ambiguous, but in the reference source, the
1023        // smoothed value is NOT fed back into later fixed gain smoothing.
1024        synth_fixed_gain = fixed_gain_smooth(p, p->lsf_q[subframe],
1025                                             p->lsf_avg, p->cur_frame_mode);
1026
1027        synth_fixed_vector = anti_sparseness(p, &fixed_sparse, p->fixed_vector,
1028                                             synth_fixed_gain, spare_vector);
1029
1030        if (synthesis(p, p->lpc[subframe], synth_fixed_gain,
1031                      synth_fixed_vector, &p->samples_in[LP_FILTER_ORDER], 0))
1032            // overflow detected -> rerun synthesis scaling pitch vector down
1033            // by a factor of 4, skipping pitch vector contribution emphasis
1034            // and adaptive gain control
1035            synthesis(p, p->lpc[subframe], synth_fixed_gain,
1036                      synth_fixed_vector, &p->samples_in[LP_FILTER_ORDER], 1);
1037
1038        postfilter(p, p->lpc[subframe], buf_out + subframe * AMR_SUBFRAME_SIZE);
1039
1040        // update buffers and history
1041        ff_clear_fixed_vector(p->fixed_vector, &fixed_sparse, AMR_SUBFRAME_SIZE);
1042        update_state(p);
1043    }
1044
1045    ff_acelp_apply_order_2_transfer_function(buf_out, buf_out, highpass_zeros,
1046                                             highpass_poles,
1047                                             highpass_gain * AMR_SAMPLE_SCALE,
1048                                             p->high_pass_mem, AMR_BLOCK_SIZE);
1049
1050    /* Update averaged lsf vector (used for fixed gain smoothing).
1051     *
1052     * Note that lsf_avg should not incorporate the current frame's LSFs
1053     * for fixed_gain_smooth.
1054     * The specification has an incorrect formula: the reference decoder uses
1055     * qbar(n-1) rather than qbar(n) in section 6.1(4) equation 71. */
1056    ff_weighted_vector_sumf(p->lsf_avg, p->lsf_avg, p->lsf_q[3],
1057                            0.84, 0.16, LP_FILTER_ORDER);
1058
1059    /* report how many samples we got */
1060    *data_size = AMR_BLOCK_SIZE * sizeof(float);
1061
1062    /* return the amount of bytes consumed if everything was OK */
1063    return frame_sizes_nb[p->cur_frame_mode] + 1; // +7 for rounding and +8 for TOC
1064}
1065
1066
1067AVCodec amrnb_decoder = {
1068    .name           = "amrnb",
1069    .type           = AVMEDIA_TYPE_AUDIO,
1070    .id             = CODEC_ID_AMR_NB,
1071    .priv_data_size = sizeof(AMRContext),
1072    .init           = amrnb_decode_init,
1073    .decode         = amrnb_decode_frame,
1074    .long_name      = NULL_IF_CONFIG_SMALL("Adaptive Multi-Rate NarrowBand"),
1075    .sample_fmts    = (enum SampleFormat[]){SAMPLE_FMT_FLT,SAMPLE_FMT_NONE},
1076};
1077