FFmpeg coverage


Directory: ../../../ffmpeg/
File: src/libavcodec/amrwbdec.c
Date: 2022-11-28 23:49:43
Exec Total Coverage
Lines: 497 519 95.8%
Branches: 230 247 93.1%

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 "config.h"
28
29 #include "libavutil/channel_layout.h"
30 #include "libavutil/common.h"
31 #include "libavutil/lfg.h"
32
33 #include "avcodec.h"
34 #include "lsp.h"
35 #include "celp_filters.h"
36 #include "celp_math.h"
37 #include "acelp_filters.h"
38 #include "acelp_vectors.h"
39 #include "acelp_pitch_delay.h"
40 #include "codec_internal.h"
41 #include "decode.h"
42
43 #define AMR_USE_16BIT_TABLES
44 #include "amr.h"
45
46 #include "amrwbdata.h"
47 #if ARCH_MIPS
48 #include "mips/amrwbdec_mips.h"
49 #endif /* ARCH_MIPS */
50
51 typedef struct AMRWBContext {
52 AMRWBFrame frame; ///< AMRWB parameters decoded from bitstream
53 enum Mode fr_cur_mode; ///< mode index of current frame
54 uint8_t fr_quality; ///< frame quality index (FQI)
55 float isf_cur[LP_ORDER]; ///< working ISF vector from current frame
56 float isf_q_past[LP_ORDER]; ///< quantized ISF vector of the previous frame
57 float isf_past_final[LP_ORDER]; ///< final processed ISF vector of the previous frame
58 double isp[4][LP_ORDER]; ///< ISP vectors from current frame
59 double isp_sub4_past[LP_ORDER]; ///< ISP vector for the 4th subframe of the previous frame
60
61 float lp_coef[4][LP_ORDER]; ///< Linear Prediction Coefficients from ISP vector
62
63 uint8_t base_pitch_lag; ///< integer part of pitch lag for the next relative subframe
64 uint8_t pitch_lag_int; ///< integer part of pitch lag of the previous subframe
65
66 float excitation_buf[AMRWB_P_DELAY_MAX + LP_ORDER + 2 + AMRWB_SFR_SIZE]; ///< current excitation and all necessary excitation history
67 float *excitation; ///< points to current excitation in excitation_buf[]
68
69 float pitch_vector[AMRWB_SFR_SIZE]; ///< adaptive codebook (pitch) vector for current subframe
70 float fixed_vector[AMRWB_SFR_SIZE]; ///< algebraic codebook (fixed) vector for current subframe
71
72 float prediction_error[4]; ///< quantified prediction errors {20log10(^gamma_gc)} for previous four subframes
73 float pitch_gain[6]; ///< quantified pitch gains for the current and previous five subframes
74 float fixed_gain[2]; ///< quantified fixed gains for the current and previous subframes
75
76 float tilt_coef; ///< {beta_1} related to the voicing of the previous subframe
77
78 float prev_sparse_fixed_gain; ///< previous fixed gain; used by anti-sparseness to determine "onset"
79 uint8_t prev_ir_filter_nr; ///< previous impulse response filter "impNr": 0 - strong, 1 - medium, 2 - none
80 float prev_tr_gain; ///< previous initial gain used by noise enhancer for threshold
81
82 float samples_az[LP_ORDER + AMRWB_SFR_SIZE]; ///< low-band samples and memory from synthesis at 12.8kHz
83 float samples_up[UPS_MEM_SIZE + AMRWB_SFR_SIZE]; ///< low-band samples and memory processed for upsampling
84 float samples_hb[LP_ORDER_16k + AMRWB_SFR_SIZE_16k]; ///< high-band samples and memory from synthesis at 16kHz
85
86 float hpf_31_mem[2], hpf_400_mem[2]; ///< previous values in the high pass filters
87 float demph_mem[1]; ///< previous value in the de-emphasis filter
88 float bpf_6_7_mem[HB_FIR_SIZE]; ///< previous values in the high-band band pass filter
89 float lpf_7_mem[HB_FIR_SIZE]; ///< previous values in the high-band low pass filter
90
91 AVLFG prng; ///< random number generator for white noise excitation
92 uint8_t first_frame; ///< flag active during decoding of the first frame
93 ACELPFContext acelpf_ctx; ///< context for filters for ACELP-based codecs
94 ACELPVContext acelpv_ctx; ///< context for vector operations for ACELP-based codecs
95 CELPFContext celpf_ctx; ///< context for filters for CELP-based codecs
96 CELPMContext celpm_ctx; ///< context for fixed point math operations
97
98 } AMRWBContext;
99
100 typedef struct AMRWBChannelsContext {
101 AMRWBContext ch[2];
102 } AMRWBChannelsContext;
103
104 22 static av_cold int amrwb_decode_init(AVCodecContext *avctx)
105 {
106 22 AMRWBChannelsContext *s = avctx->priv_data;
107 int i;
108
109
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22 if (avctx->ch_layout.nb_channels > 2) {
110 avpriv_report_missing_feature(avctx, ">2 channel AMR");
111 return AVERROR_PATCHWELCOME;
112 }
113
114
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22 if (!avctx->ch_layout.nb_channels) {
115 av_channel_layout_uninit(&avctx->ch_layout);
116 avctx->ch_layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_MONO;
117 }
118
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22 if (!avctx->sample_rate)
119 avctx->sample_rate = 16000;
120 22 avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
121
122
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44 for (int ch = 0; ch < avctx->ch_layout.nb_channels; ch++) {
123 22 AMRWBContext *ctx = &s->ch[ch];
124
125 22 av_lfg_init(&ctx->prng, 1);
126
127 22 ctx->excitation = &ctx->excitation_buf[AMRWB_P_DELAY_MAX + LP_ORDER + 1];
128 22 ctx->first_frame = 1;
129
130
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374 for (i = 0; i < LP_ORDER; i++)
131 352 ctx->isf_past_final[i] = isf_init[i] * (1.0f / (1 << 15));
132
133
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110 for (i = 0; i < 4; i++)
134 88 ctx->prediction_error[i] = MIN_ENERGY;
135
136 22 ff_acelp_filter_init(&ctx->acelpf_ctx);
137 22 ff_acelp_vectors_init(&ctx->acelpv_ctx);
138 22 ff_celp_filter_init(&ctx->celpf_ctx);
139 22 ff_celp_math_init(&ctx->celpm_ctx);
140 }
141
142 22 return 0;
143 }
144
145 /**
146 * Decode the frame header in the "MIME/storage" format. This format
147 * is simpler and does not carry the auxiliary frame information.
148 *
149 * @param[in] ctx The Context
150 * @param[in] buf Pointer to the input buffer
151 *
152 * @return The decoded header length in bytes
153 */
154 6268 static int decode_mime_header(AMRWBContext *ctx, const uint8_t *buf)
155 {
156 /* Decode frame header (1st octet) */
157 6268 ctx->fr_cur_mode = buf[0] >> 3 & 0x0F;
158 6268 ctx->fr_quality = (buf[0] & 0x4) == 0x4;
159
160 6268 return 1;
161 }
162
163 /**
164 * Decode quantized ISF vectors using 36-bit indexes (6K60 mode only).
165 *
166 * @param[in] ind Array of 5 indexes
167 * @param[out] isf_q Buffer for isf_q[LP_ORDER]
168 */
169 513 static void decode_isf_indices_36b(uint16_t *ind, float *isf_q)
170 {
171 int i;
172
173
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5130 for (i = 0; i < 9; i++)
174 4617 isf_q[i] = dico1_isf[ind[0]][i] * (1.0f / (1 << 15));
175
176
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4104 for (i = 0; i < 7; i++)
177 3591 isf_q[i + 9] = dico2_isf[ind[1]][i] * (1.0f / (1 << 15));
178
179
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3078 for (i = 0; i < 5; i++)
180 2565 isf_q[i] += dico21_isf_36b[ind[2]][i] * (1.0f / (1 << 15));
181
182
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2565 for (i = 0; i < 4; i++)
183 2052 isf_q[i + 5] += dico22_isf_36b[ind[3]][i] * (1.0f / (1 << 15));
184
185
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4104 for (i = 0; i < 7; i++)
186 3591 isf_q[i + 9] += dico23_isf_36b[ind[4]][i] * (1.0f / (1 << 15));
187 513 }
188
189 /**
190 * Decode quantized ISF vectors using 46-bit indexes (except 6K60 mode).
191 *
192 * @param[in] ind Array of 7 indexes
193 * @param[out] isf_q Buffer for isf_q[LP_ORDER]
194 */
195 5755 static void decode_isf_indices_46b(uint16_t *ind, float *isf_q)
196 {
197 int i;
198
199
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57550 for (i = 0; i < 9; i++)
200 51795 isf_q[i] = dico1_isf[ind[0]][i] * (1.0f / (1 << 15));
201
202
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46040 for (i = 0; i < 7; i++)
203 40285 isf_q[i + 9] = dico2_isf[ind[1]][i] * (1.0f / (1 << 15));
204
205
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23020 for (i = 0; i < 3; i++)
206 17265 isf_q[i] += dico21_isf[ind[2]][i] * (1.0f / (1 << 15));
207
208
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23020 for (i = 0; i < 3; i++)
209 17265 isf_q[i + 3] += dico22_isf[ind[3]][i] * (1.0f / (1 << 15));
210
211
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23020 for (i = 0; i < 3; i++)
212 17265 isf_q[i + 6] += dico23_isf[ind[4]][i] * (1.0f / (1 << 15));
213
214
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23020 for (i = 0; i < 3; i++)
215 17265 isf_q[i + 9] += dico24_isf[ind[5]][i] * (1.0f / (1 << 15));
216
217
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28775 for (i = 0; i < 4; i++)
218 23020 isf_q[i + 12] += dico25_isf[ind[6]][i] * (1.0f / (1 << 15));
219 5755 }
220
221 /**
222 * Apply mean and past ISF values using the prediction factor.
223 * Updates past ISF vector.
224 *
225 * @param[in,out] isf_q Current quantized ISF
226 * @param[in,out] isf_past Past quantized ISF
227 */
228 6268 static void isf_add_mean_and_past(float *isf_q, float *isf_past)
229 {
230 int i;
231 float tmp;
232
233
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106556 for (i = 0; i < LP_ORDER; i++) {
234 100288 tmp = isf_q[i];
235 100288 isf_q[i] += isf_mean[i] * (1.0f / (1 << 15));
236 100288 isf_q[i] += PRED_FACTOR * isf_past[i];
237 100288 isf_past[i] = tmp;
238 }
239 6268 }
240
241 /**
242 * Interpolate the fourth ISP vector from current and past frames
243 * to obtain an ISP vector for each subframe.
244 *
245 * @param[in,out] isp_q ISPs for each subframe
246 * @param[in] isp4_past Past ISP for subframe 4
247 */
248 6268 static void interpolate_isp(double isp_q[4][LP_ORDER], const double *isp4_past)
249 {
250 int i, k;
251
252
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25072 for (k = 0; k < 3; k++) {
253 18804 float c = isfp_inter[k];
254
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319668 for (i = 0; i < LP_ORDER; i++)
255 300864 isp_q[k][i] = (1.0 - c) * isp4_past[i] + c * isp_q[3][i];
256 }
257 6268 }
258
259 /**
260 * Decode an adaptive codebook index into pitch lag (except 6k60, 8k85 modes).
261 * Calculate integer lag and fractional lag always using 1/4 resolution.
262 * In 1st and 3rd subframes the index is relative to last subframe integer lag.
263 *
264 * @param[out] lag_int Decoded integer pitch lag
265 * @param[out] lag_frac Decoded fractional pitch lag
266 * @param[in] pitch_index Adaptive codebook pitch index
267 * @param[in,out] base_lag_int Base integer lag used in relative subframes
268 * @param[in] subframe Current subframe index (0 to 3)
269 */
270 20968 static void decode_pitch_lag_high(int *lag_int, int *lag_frac, int pitch_index,
271 uint8_t *base_lag_int, int subframe)
272 {
273
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20968 if (subframe == 0 || subframe == 2) {
274
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10484 if (pitch_index < 376) {
275 7630 *lag_int = (pitch_index + 137) >> 2;
276 7630 *lag_frac = pitch_index - (*lag_int << 2) + 136;
277
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2854 } else if (pitch_index < 440) {
278 1248 *lag_int = (pitch_index + 257 - 376) >> 1;
279 1248 *lag_frac = (pitch_index - (*lag_int << 1) + 256 - 376) * 2;
280 /* the actual resolution is 1/2 but expressed as 1/4 */
281 } else {
282 1606 *lag_int = pitch_index - 280;
283 1606 *lag_frac = 0;
284 }
285 /* minimum lag for next subframe */
286 10484 *base_lag_int = av_clip(*lag_int - 8 - (*lag_frac < 0),
287 AMRWB_P_DELAY_MIN, AMRWB_P_DELAY_MAX - 15);
288 // XXX: the spec states clearly that *base_lag_int should be
289 // the nearest integer to *lag_int (minus 8), but the ref code
290 // actually always uses its floor, I'm following the latter
291 } else {
292 10484 *lag_int = (pitch_index + 1) >> 2;
293 10484 *lag_frac = pitch_index - (*lag_int << 2);
294 10484 *lag_int += *base_lag_int;
295 }
296 20968 }
297
298 /**
299 * Decode an adaptive codebook index into pitch lag for 8k85 and 6k60 modes.
300 * The description is analogous to decode_pitch_lag_high, but in 6k60 the
301 * relative index is used for all subframes except the first.
302 */
303 4104 static void decode_pitch_lag_low(int *lag_int, int *lag_frac, int pitch_index,
304 uint8_t *base_lag_int, int subframe, enum Mode mode)
305 {
306
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4104 if (subframe == 0 || (subframe == 2 && mode != MODE_6k60)) {
307
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1539 if (pitch_index < 116) {
308 850 *lag_int = (pitch_index + 69) >> 1;
309 850 *lag_frac = (pitch_index - (*lag_int << 1) + 68) * 2;
310 } else {
311 689 *lag_int = pitch_index - 24;
312 689 *lag_frac = 0;
313 }
314 // XXX: same problem as before
315 1539 *base_lag_int = av_clip(*lag_int - 8 - (*lag_frac < 0),
316 AMRWB_P_DELAY_MIN, AMRWB_P_DELAY_MAX - 15);
317 } else {
318 2565 *lag_int = (pitch_index + 1) >> 1;
319 2565 *lag_frac = (pitch_index - (*lag_int << 1)) * 2;
320 2565 *lag_int += *base_lag_int;
321 }
322 4104 }
323
324 /**
325 * Find the pitch vector by interpolating the past excitation at the
326 * pitch delay, which is obtained in this function.
327 *
328 * @param[in,out] ctx The context
329 * @param[in] amr_subframe Current subframe data
330 * @param[in] subframe Current subframe index (0 to 3)
331 */
332 25072 static void decode_pitch_vector(AMRWBContext *ctx,
333 const AMRWBSubFrame *amr_subframe,
334 const int subframe)
335 {
336 int pitch_lag_int, pitch_lag_frac;
337 int i;
338 25072 float *exc = ctx->excitation;
339 25072 enum Mode mode = ctx->fr_cur_mode;
340
341
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25072 if (mode <= MODE_8k85) {
342 4104 decode_pitch_lag_low(&pitch_lag_int, &pitch_lag_frac, amr_subframe->adap,
343 &ctx->base_pitch_lag, subframe, mode);
344 } else
345 20968 decode_pitch_lag_high(&pitch_lag_int, &pitch_lag_frac, amr_subframe->adap,
346 &ctx->base_pitch_lag, subframe);
347
348 25072 ctx->pitch_lag_int = pitch_lag_int;
349 25072 pitch_lag_int += pitch_lag_frac > 0;
350
351 /* Calculate the pitch vector by interpolating the past excitation at the
352 pitch lag using a hamming windowed sinc function */
353 50144 ctx->acelpf_ctx.acelp_interpolatef(exc,
354 25072 exc + 1 - pitch_lag_int,
355 ac_inter, 4,
356
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25072 pitch_lag_frac + (pitch_lag_frac > 0 ? 0 : 4),
357 LP_ORDER, AMRWB_SFR_SIZE + 1);
358
359 /* Check which pitch signal path should be used
360 * 6k60 and 8k85 modes have the ltp flag set to 0 */
361
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25072 if (amr_subframe->ltp) {
362 7140 memcpy(ctx->pitch_vector, exc, AMRWB_SFR_SIZE * sizeof(float));
363 } else {
364
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1165580 for (i = 0; i < AMRWB_SFR_SIZE; i++)
365 1147648 ctx->pitch_vector[i] = 0.18 * exc[i - 1] + 0.64 * exc[i] +
366 1147648 0.18 * exc[i + 1];
367 17932 memcpy(exc, ctx->pitch_vector, AMRWB_SFR_SIZE * sizeof(float));
368 }
369 25072 }
370
371 /** Get x bits in the index interval [lsb,lsb+len-1] inclusive */
372 #define BIT_STR(x,lsb,len) av_mod_uintp2((x) >> (lsb), (len))
373
374 /** Get the bit at specified position */
375 #define BIT_POS(x, p) (((x) >> (p)) & 1)
376
377 /**
378 * The next six functions decode_[i]p_track decode exactly i pulses
379 * positions and amplitudes (-1 or 1) in a subframe track using
380 * an encoded pulse indexing (TS 26.190 section 5.8.2).
381 *
382 * The results are given in out[], in which a negative number means
383 * amplitude -1 and vice versa (i.e., ampl(x) = x / abs(x) ).
384 *
385 * @param[out] out Output buffer (writes i elements)
386 * @param[in] code Pulse index (no. of bits varies, see below)
387 * @param[in] m (log2) Number of potential positions
388 * @param[in] off Offset for decoded positions
389 */
390 106180 static inline void decode_1p_track(int *out, int code, int m, int off)
391 {
392 106180 int pos = BIT_STR(code, 0, m) + off; ///code: m+1 bits
393
394
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106180 out[0] = BIT_POS(code, m) ? -pos : pos;
395 106180 }
396
397 147226 static inline void decode_2p_track(int *out, int code, int m, int off) ///code: 2m+1 bits
398 {
399 147226 int pos0 = BIT_STR(code, m, m) + off;
400 147226 int pos1 = BIT_STR(code, 0, m) + off;
401
402
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147226 out[0] = BIT_POS(code, 2*m) ? -pos0 : pos0;
403
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147226 out[1] = BIT_POS(code, 2*m) ? -pos1 : pos1;
404
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147226 out[1] = pos0 > pos1 ? -out[1] : out[1];
405 147226 }
406
407 68748 static void decode_3p_track(int *out, int code, int m, int off) ///code: 3m+1 bits
408 {
409 68748 int half_2p = BIT_POS(code, 2*m - 1) << (m - 1);
410
411 68748 decode_2p_track(out, BIT_STR(code, 0, 2*m - 1),
412 m - 1, off + half_2p);
413 68748 decode_1p_track(out + 2, BIT_STR(code, 2*m, m + 1), m, off);
414 68748 }
415
416 32563 static void decode_4p_track(int *out, int code, int m, int off) ///code: 4m bits
417 {
418 int half_4p, subhalf_2p;
419 32563 int b_offset = 1 << (m - 1);
420
421
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32563 switch (BIT_STR(code, 4*m - 2, 2)) { /* case ID (2 bits) */
422 3612 case 0: /* 0 pulses in A, 4 pulses in B or vice versa */
423 3612 half_4p = BIT_POS(code, 4*m - 3) << (m - 1); // which has 4 pulses
424 3612 subhalf_2p = BIT_POS(code, 2*m - 3) << (m - 2);
425
426 3612 decode_2p_track(out, BIT_STR(code, 0, 2*m - 3),
427 3612 m - 2, off + half_4p + subhalf_2p);
428 3612 decode_2p_track(out + 2, BIT_STR(code, 2*m - 2, 2*m - 1),
429 m - 1, off + half_4p);
430 3612 break;
431 7991 case 1: /* 1 pulse in A, 3 pulses in B */
432 7991 decode_1p_track(out, BIT_STR(code, 3*m - 2, m),
433 m - 1, off);
434 7991 decode_3p_track(out + 1, BIT_STR(code, 0, 3*m - 2),
435 m - 1, off + b_offset);
436 7991 break;
437 12806 case 2: /* 2 pulses in each half */
438 12806 decode_2p_track(out, BIT_STR(code, 2*m - 1, 2*m - 1),
439 m - 1, off);
440 12806 decode_2p_track(out + 2, BIT_STR(code, 0, 2*m - 1),
441 m - 1, off + b_offset);
442 12806 break;
443 8154 case 3: /* 3 pulses in A, 1 pulse in B */
444 8154 decode_3p_track(out, BIT_STR(code, m, 3*m - 2),
445 m - 1, off);
446 8154 decode_1p_track(out + 3, BIT_STR(code, 0, m),
447 m - 1, off + b_offset);
448 8154 break;
449 }
450 32563 }
451
452 13079 static void decode_5p_track(int *out, int code, int m, int off) ///code: 5m bits
453 {
454 13079 int half_3p = BIT_POS(code, 5*m - 1) << (m - 1);
455
456 13079 decode_3p_track(out, BIT_STR(code, 2*m + 1, 3*m - 2),
457 m - 1, off + half_3p);
458
459 13079 decode_2p_track(out + 3, BIT_STR(code, 0, 2*m + 1), m, off);
460 13079 }
461
462 42832 static void decode_6p_track(int *out, int code, int m, int off) ///code: 6m-2 bits
463 {
464 42832 int b_offset = 1 << (m - 1);
465 /* which half has more pulses in cases 0 to 2 */
466 42832 int half_more = BIT_POS(code, 6*m - 5) << (m - 1);
467 42832 int half_other = b_offset - half_more;
468
469
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42832 switch (BIT_STR(code, 6*m - 4, 2)) { /* case ID (2 bits) */
470 1297 case 0: /* 0 pulses in A, 6 pulses in B or vice versa */
471 1297 decode_1p_track(out, BIT_STR(code, 0, m),
472 m - 1, off + half_more);
473 1297 decode_5p_track(out + 1, BIT_STR(code, m, 5*m - 5),
474 m - 1, off + half_more);
475 1297 break;
476 7678 case 1: /* 1 pulse in A, 5 pulses in B or vice versa */
477 7678 decode_1p_track(out, BIT_STR(code, 0, m),
478 m - 1, off + half_other);
479 7678 decode_5p_track(out + 1, BIT_STR(code, m, 5*m - 5),
480 m - 1, off + half_more);
481 7678 break;
482 20251 case 2: /* 2 pulses in A, 4 pulses in B or vice versa */
483 20251 decode_2p_track(out, BIT_STR(code, 0, 2*m - 1),
484 m - 1, off + half_other);
485 20251 decode_4p_track(out + 2, BIT_STR(code, 2*m - 1, 4*m - 4),
486 m - 1, off + half_more);
487 20251 break;
488 13606 case 3: /* 3 pulses in A, 3 pulses in B */
489 13606 decode_3p_track(out, BIT_STR(code, 3*m - 2, 3*m - 2),
490 m - 1, off);
491 13606 decode_3p_track(out + 3, BIT_STR(code, 0, 3*m - 2),
492 m - 1, off + b_offset);
493 13606 break;
494 }
495 42832 }
496
497 /**
498 * Decode the algebraic codebook index to pulse positions and signs,
499 * then construct the algebraic codebook vector.
500 *
501 * @param[out] fixed_vector Buffer for the fixed codebook excitation
502 * @param[in] pulse_hi MSBs part of the pulse index array (higher modes only)
503 * @param[in] pulse_lo LSBs part of the pulse index array
504 * @param[in] mode Mode of the current frame
505 */
506 25072 static void decode_fixed_vector(float *fixed_vector, const uint16_t *pulse_hi,
507 const uint16_t *pulse_lo, const enum Mode mode)
508 {
509 /* sig_pos stores for each track the decoded pulse position indexes
510 * (1-based) multiplied by its corresponding amplitude (+1 or -1) */
511 int sig_pos[4][6];
512
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25072 int spacing = (mode == MODE_6k60) ? 2 : 4;
513 int i, j;
514
515
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25072 switch (mode) {
516 2052 case MODE_6k60:
517
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6156 for (i = 0; i < 2; i++)
518 4104 decode_1p_track(sig_pos[i], pulse_lo[i], 5, 1);
519 2052 break;
520 2052 case MODE_8k85:
521
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10260 for (i = 0; i < 4; i++)
522 8208 decode_1p_track(sig_pos[i], pulse_lo[i], 4, 1);
523 2052 break;
524 2052 case MODE_12k65:
525
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10260 for (i = 0; i < 4; i++)
526 8208 decode_2p_track(sig_pos[i], pulse_lo[i], 4, 1);
527 2052 break;
528 2052 case MODE_14k25:
529
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6156 for (i = 0; i < 2; i++)
530 4104 decode_3p_track(sig_pos[i], pulse_lo[i], 4, 1);
531
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6156 for (i = 2; i < 4; i++)
532 4104 decode_2p_track(sig_pos[i], pulse_lo[i], 4, 1);
533 2052 break;
534 2052 case MODE_15k85:
535
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10260 for (i = 0; i < 4; i++)
536 8208 decode_3p_track(sig_pos[i], pulse_lo[i], 4, 1);
537 2052 break;
538 2052 case MODE_18k25:
539
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10260 for (i = 0; i < 4; i++)
540 8208 decode_4p_track(sig_pos[i], (int) pulse_lo[i] +
541 8208 ((int) pulse_hi[i] << 14), 4, 1);
542 2052 break;
543 2052 case MODE_19k85:
544
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6156 for (i = 0; i < 2; i++)
545 4104 decode_5p_track(sig_pos[i], (int) pulse_lo[i] +
546 4104 ((int) pulse_hi[i] << 10), 4, 1);
547
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6156 for (i = 2; i < 4; i++)
548 4104 decode_4p_track(sig_pos[i], (int) pulse_lo[i] +
549 4104 ((int) pulse_hi[i] << 14), 4, 1);
550 2052 break;
551 10708 case MODE_23k05:
552 case MODE_23k85:
553
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53540 for (i = 0; i < 4; i++)
554 42832 decode_6p_track(sig_pos[i], (int) pulse_lo[i] +
555 42832 ((int) pulse_hi[i] << 11), 4, 1);
556 10708 break;
557 }
558
559 25072 memset(fixed_vector, 0, sizeof(float) * AMRWB_SFR_SIZE);
560
561
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562
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500920 for (j = 0; j < pulses_nb_per_mode_tr[mode][i]; j++) {
563 400632 int pos = (FFABS(sig_pos[i][j]) - 1) * spacing + i;
564
565
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400632 fixed_vector[pos] += sig_pos[i][j] < 0 ? -1.0 : 1.0;
566 }
567 25072 }
568
569 /**
570 * Decode pitch gain and fixed gain correction factor.
571 *
572 * @param[in] vq_gain Vector-quantized index for gains
573 * @param[in] mode Mode of the current frame
574 * @param[out] fixed_gain_factor Decoded fixed gain correction factor
575 * @param[out] pitch_gain Decoded pitch gain
576 */
577 25072 static void decode_gains(const uint8_t vq_gain, const enum Mode mode,
578 float *fixed_gain_factor, float *pitch_gain)
579 {
580
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25072 const int16_t *gains = (mode <= MODE_8k85 ? qua_gain_6b[vq_gain] :
581 20968 qua_gain_7b[vq_gain]);
582
583 25072 *pitch_gain = gains[0] * (1.0f / (1 << 14));
584 25072 *fixed_gain_factor = gains[1] * (1.0f / (1 << 11));
585 25072 }
586
587 /**
588 * Apply pitch sharpening filters to the fixed codebook vector.
589 *
590 * @param[in] ctx The context
591 * @param[in,out] fixed_vector Fixed codebook excitation
592 */
593 // XXX: Spec states this procedure should be applied when the pitch
594 // lag is less than 64, but this checking seems absent in reference and AMR-NB
595 25072 static void pitch_sharpening(AMRWBContext *ctx, float *fixed_vector)
596 {
597 int i;
598
599 /* Tilt part */
600
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1604608 for (i = AMRWB_SFR_SIZE - 1; i != 0; i--)
601 1579536 fixed_vector[i] -= fixed_vector[i - 1] * ctx->tilt_coef;
602
603 /* Periodicity enhancement part */
604
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187025 for (i = ctx->pitch_lag_int; i < AMRWB_SFR_SIZE; i++)
605 161953 fixed_vector[i] += fixed_vector[i - ctx->pitch_lag_int] * 0.85;
606 25072 }
607
608 /**
609 * Calculate the voicing factor (-1.0 = unvoiced to 1.0 = voiced).
610 *
611 * @param[in] p_vector, f_vector Pitch and fixed excitation vectors
612 * @param[in] p_gain, f_gain Pitch and fixed gains
613 * @param[in] ctx The context
614 */
615 // XXX: There is something wrong with the precision here! The magnitudes
616 // of the energies are not correct. Please check the reference code carefully
617 25072 static float voice_factor(float *p_vector, float p_gain,
618 float *f_vector, float f_gain,
619 CELPMContext *ctx)
620 {
621 25072 double p_ener = (double) ctx->dot_productf(p_vector, p_vector,
622 25072 AMRWB_SFR_SIZE) *
623 25072 p_gain * p_gain;
624 25072 double f_ener = (double) ctx->dot_productf(f_vector, f_vector,
625 25072 AMRWB_SFR_SIZE) *
626 25072 f_gain * f_gain;
627
628 25072 return (p_ener - f_ener) / (p_ener + f_ener + 0.01);
629 }
630
631 /**
632 * Reduce fixed vector sparseness by smoothing with one of three IR filters,
633 * also known as "adaptive phase dispersion".
634 *
635 * @param[in] ctx The context
636 * @param[in,out] fixed_vector Unfiltered fixed vector
637 * @param[out] buf Space for modified vector if necessary
638 *
639 * @return The potentially overwritten filtered fixed vector address
640 */
641 25072 static float *anti_sparseness(AMRWBContext *ctx,
642 float *fixed_vector, float *buf)
643 {
644 int ir_filter_nr;
645
646
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25072 if (ctx->fr_cur_mode > MODE_8k85) // no filtering in higher modes
647 20968 return fixed_vector;
648
649
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4104 if (ctx->pitch_gain[0] < 0.6) {
650 2261 ir_filter_nr = 0; // strong filtering
651
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1843 } else if (ctx->pitch_gain[0] < 0.9) {
652 712 ir_filter_nr = 1; // medium filtering
653 } else
654 1131 ir_filter_nr = 2; // no filtering
655
656 /* detect 'onset' */
657
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4104 if (ctx->fixed_gain[0] > 3.0 * ctx->fixed_gain[1]) {
658
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61 if (ir_filter_nr < 2)
659 39 ir_filter_nr++;
660 } else {
661 4043 int i, count = 0;
662
663
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28301 for (i = 0; i < 6; i++)
664
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24258 if (ctx->pitch_gain[i] < 0.6)
665 13373 count++;
666
667
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4043 if (count > 2)
668 2724 ir_filter_nr = 0;
669
670
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4043 if (ir_filter_nr > ctx->prev_ir_filter_nr + 1)
671 83 ir_filter_nr--;
672 }
673
674 /* update ir filter strength history */
675 4104 ctx->prev_ir_filter_nr = ir_filter_nr;
676
677 4104 ir_filter_nr += (ctx->fr_cur_mode == MODE_8k85);
678
679
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4104 if (ir_filter_nr < 2) {
680 int i;
681 3197 const float *coef = ir_filters_lookup[ir_filter_nr];
682
683 /* Circular convolution code in the reference
684 * decoder was modified to avoid using one
685 * extra array. The filtered vector is given by:
686 *
687 * c2(n) = sum(i,0,len-1){ c(i) * coef( (n - i + len) % len ) }
688 */
689
690 3197 memset(buf, 0, sizeof(float) * AMRWB_SFR_SIZE);
691
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207805 for (i = 0; i < AMRWB_SFR_SIZE; i++)
692
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204608 if (fixed_vector[i])
693 21471 ff_celp_circ_addf(buf, buf, coef, i, fixed_vector[i],
694 AMRWB_SFR_SIZE);
695 3197 fixed_vector = buf;
696 }
697
698 4104 return fixed_vector;
699 }
700
701 /**
702 * Calculate a stability factor {teta} based on distance between
703 * current and past isf. A value of 1 shows maximum signal stability.
704 */
705 6268 static float stability_factor(const float *isf, const float *isf_past)
706 {
707 int i;
708 6268 float acc = 0.0;
709
710
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100288 for (i = 0; i < LP_ORDER - 1; i++)
711 94020 acc += (isf[i] - isf_past[i]) * (isf[i] - isf_past[i]);
712
713 // XXX: This part is not so clear from the reference code
714 // the result is more accurate changing the "/ 256" to "* 512"
715
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6268 return FFMAX(0.0, 1.25 - acc * 0.8 * 512);
716 }
717
718 /**
719 * Apply a non-linear fixed gain smoothing in order to reduce
720 * fluctuation in the energy of excitation.
721 *
722 * @param[in] fixed_gain Unsmoothed fixed gain
723 * @param[in,out] prev_tr_gain Previous threshold gain (updated)
724 * @param[in] voice_fac Frame voicing factor
725 * @param[in] stab_fac Frame stability factor
726 *
727 * @return The smoothed gain
728 */
729 25072 static float noise_enhancer(float fixed_gain, float *prev_tr_gain,
730 float voice_fac, float stab_fac)
731 {
732 25072 float sm_fac = 0.5 * (1 - voice_fac) * stab_fac;
733 float g0;
734
735 // XXX: the following fixed-point constants used to in(de)crement
736 // gain by 1.5dB were taken from the reference code, maybe it could
737 // be simpler
738
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25072 if (fixed_gain < *prev_tr_gain) {
739
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13115 g0 = FFMIN(*prev_tr_gain, fixed_gain + fixed_gain *
740 (6226 * (1.0f / (1 << 15)))); // +1.5 dB
741 } else
742
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11957 g0 = FFMAX(*prev_tr_gain, fixed_gain *
743 (27536 * (1.0f / (1 << 15)))); // -1.5 dB
744
745 25072 *prev_tr_gain = g0; // update next frame threshold
746
747 25072 return sm_fac * g0 + (1 - sm_fac) * fixed_gain;
748 }
749
750 /**
751 * Filter the fixed_vector to emphasize the higher frequencies.
752 *
753 * @param[in,out] fixed_vector Fixed codebook vector
754 * @param[in] voice_fac Frame voicing factor
755 */
756 25072 static void pitch_enhancer(float *fixed_vector, float voice_fac)
757 {
758 int i;
759 25072 float cpe = 0.125 * (1 + voice_fac);
760 25072 float last = fixed_vector[0]; // holds c(i - 1)
761
762 25072 fixed_vector[0] -= cpe * fixed_vector[1];
763
764
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1579536 for (i = 1; i < AMRWB_SFR_SIZE - 1; i++) {
765 1554464 float cur = fixed_vector[i];
766
767 1554464 fixed_vector[i] -= cpe * (last + fixed_vector[i + 1]);
768 1554464 last = cur;
769 }
770
771 25072 fixed_vector[AMRWB_SFR_SIZE - 1] -= cpe * last;
772 25072 }
773
774 /**
775 * Conduct 16th order linear predictive coding synthesis from excitation.
776 *
777 * @param[in] ctx Pointer to the AMRWBContext
778 * @param[in] lpc Pointer to the LPC coefficients
779 * @param[out] excitation Buffer for synthesis final excitation
780 * @param[in] fixed_gain Fixed codebook gain for synthesis
781 * @param[in] fixed_vector Algebraic codebook vector
782 * @param[in,out] samples Pointer to the output samples and memory
783 */
784 25072 static void synthesis(AMRWBContext *ctx, float *lpc, float *excitation,
785 float fixed_gain, const float *fixed_vector,
786 float *samples)
787 {
788 25072 ctx->acelpv_ctx.weighted_vector_sumf(excitation, ctx->pitch_vector, fixed_vector,
789 ctx->pitch_gain[0], fixed_gain, AMRWB_SFR_SIZE);
790
791 /* emphasize pitch vector contribution in low bitrate modes */
792
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25072 if (ctx->pitch_gain[0] > 0.5 && ctx->fr_cur_mode <= MODE_8k85) {
793 int i;
794 1897 float energy = ctx->celpm_ctx.dot_productf(excitation, excitation,
795 AMRWB_SFR_SIZE);
796
797 // XXX: Weird part in both ref code and spec. A unknown parameter
798 // {beta} seems to be identical to the current pitch gain
799 1897 float pitch_factor = 0.25 * ctx->pitch_gain[0] * ctx->pitch_gain[0];
800
801
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123305 for (i = 0; i < AMRWB_SFR_SIZE; i++)
802 121408 excitation[i] += pitch_factor * ctx->pitch_vector[i];
803
804 1897 ff_scale_vector_to_given_sum_of_squares(excitation, excitation,
805 energy, AMRWB_SFR_SIZE);
806 }
807
808 25072 ctx->celpf_ctx.celp_lp_synthesis_filterf(samples, lpc, excitation,
809 AMRWB_SFR_SIZE, LP_ORDER);
810 25072 }
811
812 /**
813 * Apply to synthesis a de-emphasis filter of the form:
814 * H(z) = 1 / (1 - m * z^-1)
815 *
816 * @param[out] out Output buffer
817 * @param[in] in Input samples array with in[-1]
818 * @param[in] m Filter coefficient
819 * @param[in,out] mem State from last filtering
820 */
821 25072 static void de_emphasis(float *out, float *in, float m, float mem[1])
822 {
823 int i;
824
825 25072 out[0] = in[0] + m * mem[0];
826
827
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1604608 for (i = 1; i < AMRWB_SFR_SIZE; i++)
828 1579536 out[i] = in[i] + out[i - 1] * m;
829
830 25072 mem[0] = out[AMRWB_SFR_SIZE - 1];
831 25072 }
832
833 /**
834 * Upsample a signal by 5/4 ratio (from 12.8kHz to 16kHz) using
835 * a FIR interpolation filter. Uses past data from before *in address.
836 *
837 * @param[out] out Buffer for interpolated signal
838 * @param[in] in Current signal data (length 0.8*o_size)
839 * @param[in] o_size Output signal length
840 * @param[in] ctx The context
841 */
842 25072 static void upsample_5_4(float *out, const float *in, int o_size, CELPMContext *ctx)
843 {
844 25072 const float *in0 = in - UPS_FIR_SIZE + 1;
845 int i, j, k;
846 25072 int int_part = 0, frac_part;
847
848 25072 i = 0;
849
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426224 for (j = 0; j < o_size / 5; j++) {
850 401152 out[i] = in[int_part];
851 401152 frac_part = 4;
852 401152 i++;
853
854
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2005760 for (k = 1; k < 5; k++) {
855 3209216 out[i] = ctx->dot_productf(in0 + int_part,
856 1604608 upsample_fir[4 - frac_part],
857 UPS_MEM_SIZE);
858 1604608 int_part++;
859 1604608 frac_part--;
860 1604608 i++;
861 }
862 }
863 25072 }
864
865 /**
866 * Calculate the high-band gain based on encoded index (23k85 mode) or
867 * on the low-band speech signal and the Voice Activity Detection flag.
868 *
869 * @param[in] ctx The context
870 * @param[in] synth LB speech synthesis at 12.8k
871 * @param[in] hb_idx Gain index for mode 23k85 only
872 * @param[in] vad VAD flag for the frame
873 */
874 25072 static float find_hb_gain(AMRWBContext *ctx, const float *synth,
875 uint16_t hb_idx, uint8_t vad)
876 {
877 25072 int wsp = (vad > 0);
878 float tilt;
879 float tmp;
880
881
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25072 if (ctx->fr_cur_mode == MODE_23k85)
882 8656 return qua_hb_gain[hb_idx] * (1.0f / (1 << 14));
883
884 16416 tmp = ctx->celpm_ctx.dot_productf(synth, synth + 1, AMRWB_SFR_SIZE - 1);
885
886
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16416 if (tmp > 0) {
887 15653 tilt = tmp / ctx->celpm_ctx.dot_productf(synth, synth, AMRWB_SFR_SIZE);
888 } else
889 763 tilt = 0;
890
891 /* return gain bounded by [0.1, 1.0] */
892 16416 return av_clipf((1.0 - tilt) * (1.25 - 0.25 * wsp), 0.1, 1.0);
893 }
894
895 /**
896 * Generate the high-band excitation with the same energy from the lower
897 * one and scaled by the given gain.
898 *
899 * @param[in] ctx The context
900 * @param[out] hb_exc Buffer for the excitation
901 * @param[in] synth_exc Low-band excitation used for synthesis
902 * @param[in] hb_gain Wanted excitation gain
903 */
904 25072 static void scaled_hb_excitation(AMRWBContext *ctx, float *hb_exc,
905 const float *synth_exc, float hb_gain)
906 {
907 int i;
908 25072 float energy = ctx->celpm_ctx.dot_productf(synth_exc, synth_exc,
909 AMRWB_SFR_SIZE);
910
911 /* Generate a white-noise excitation */
912
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2030832 for (i = 0; i < AMRWB_SFR_SIZE_16k; i++)
913 2005760 hb_exc[i] = 32768.0 - (uint16_t) av_lfg_get(&ctx->prng);
914
915 25072 ff_scale_vector_to_given_sum_of_squares(hb_exc, hb_exc,
916 25072 energy * hb_gain * hb_gain,
917 AMRWB_SFR_SIZE_16k);
918 25072 }
919
920 /**
921 * Calculate the auto-correlation for the ISF difference vector.
922 */
923 6156 static float auto_correlation(float *diff_isf, float mean, int lag)
924 {
925 int i;
926 6156 float sum = 0.0;
927
928
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49248 for (i = 7; i < LP_ORDER - 2; i++) {
929 43092 float prod = (diff_isf[i] - mean) * (diff_isf[i - lag] - mean);
930 43092 sum += prod * prod;
931 }
932 6156 return sum;
933 }
934
935 /**
936 * Extrapolate a ISF vector to the 16kHz range (20th order LP)
937 * used at mode 6k60 LP filter for the high frequency band.
938 *
939 * @param[out] isf Buffer for extrapolated isf; contains LP_ORDER
940 * values on input
941 */
942 2052 static void extrapolate_isf(float isf[LP_ORDER_16k])
943 {
944 float diff_isf[LP_ORDER - 2], diff_mean;
945 float corr_lag[3];
946 float est, scale;
947 int i, j, i_max_corr;
948
949 2052 isf[LP_ORDER_16k - 1] = isf[LP_ORDER - 1];
950
951 /* Calculate the difference vector */
952
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30780 for (i = 0; i < LP_ORDER - 2; i++)
953 28728 diff_isf[i] = isf[i + 1] - isf[i];
954
955 2052 diff_mean = 0.0;
956
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26676 for (i = 2; i < LP_ORDER - 2; i++)
957 24624 diff_mean += diff_isf[i] * (1.0f / (LP_ORDER - 4));
958
959 /* Find which is the maximum autocorrelation */
960 2052 i_max_corr = 0;
961
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8208 for (i = 0; i < 3; i++) {
962 6156 corr_lag[i] = auto_correlation(diff_isf, diff_mean, i + 2);
963
964
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6156 if (corr_lag[i] > corr_lag[i_max_corr])
965 2154 i_max_corr = i;
966 }
967 2052 i_max_corr++;
968
969
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10260 for (i = LP_ORDER - 1; i < LP_ORDER_16k - 1; i++)
970 8208 isf[i] = isf[i - 1] + isf[i - 1 - i_max_corr]
971 8208 - isf[i - 2 - i_max_corr];
972
973 /* Calculate an estimate for ISF(18) and scale ISF based on the error */
974 2052 est = 7965 + (isf[2] - isf[3] - isf[4]) / 6.0;
975
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2052 scale = 0.5 * (FFMIN(est, 7600) - isf[LP_ORDER - 2]) /
976 2052 (isf[LP_ORDER_16k - 2] - isf[LP_ORDER - 2]);
977
978
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10260 for (i = LP_ORDER - 1, j = 0; i < LP_ORDER_16k - 1; i++, j++)
979 8208 diff_isf[j] = scale * (isf[i] - isf[i - 1]);
980
981 /* Stability insurance */
982
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8208 for (i = 1; i < LP_ORDER_16k - LP_ORDER; i++)
983
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6156 if (diff_isf[i] + diff_isf[i - 1] < 5.0) {
984 if (diff_isf[i] > diff_isf[i - 1]) {
985 diff_isf[i - 1] = 5.0 - diff_isf[i];
986 } else
987 diff_isf[i] = 5.0 - diff_isf[i - 1];
988 }
989
990
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10260 for (i = LP_ORDER - 1, j = 0; i < LP_ORDER_16k - 1; i++, j++)
991 8208 isf[i] = isf[i - 1] + diff_isf[j] * (1.0f / (1 << 15));
992
993 /* Scale the ISF vector for 16000 Hz */
994
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41040 for (i = 0; i < LP_ORDER_16k - 1; i++)
995 38988 isf[i] *= 0.8;
996 2052 }
997
998 /**
999 * Spectral expand the LP coefficients using the equation:
1000 * y[i] = x[i] * (gamma ** i)
1001 *
1002 * @param[out] out Output buffer (may use input array)
1003 * @param[in] lpc LP coefficients array
1004 * @param[in] gamma Weighting factor
1005 * @param[in] size LP array size
1006 */
1007 25072 static void lpc_weighting(float *out, const float *lpc, float gamma, int size)
1008 {
1009 int i;
1010 25072 float fac = gamma;
1011
1012
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434432 for (i = 0; i < size; i++) {
1013 409360 out[i] = lpc[i] * fac;
1014 409360 fac *= gamma;
1015 }
1016 25072 }
1017
1018 /**
1019 * Conduct 20th order linear predictive coding synthesis for the high
1020 * frequency band excitation at 16kHz.
1021 *
1022 * @param[in] ctx The context
1023 * @param[in] subframe Current subframe index (0 to 3)
1024 * @param[in,out] samples Pointer to the output speech samples
1025 * @param[in] exc Generated white-noise scaled excitation
1026 * @param[in] isf Current frame isf vector
1027 * @param[in] isf_past Past frame final isf vector
1028 */
1029 25072 static void hb_synthesis(AMRWBContext *ctx, int subframe, float *samples,
1030 const float *exc, const float *isf, const float *isf_past)
1031 {
1032 float hb_lpc[LP_ORDER_16k];
1033 25072 enum Mode mode = ctx->fr_cur_mode;
1034
1035
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25072 if (mode == MODE_6k60) {
1036 float e_isf[LP_ORDER_16k]; // ISF vector for extrapolation
1037 double e_isp[LP_ORDER_16k];
1038
1039 2052 ctx->acelpv_ctx.weighted_vector_sumf(e_isf, isf_past, isf, isfp_inter[subframe],
1040 2052 1.0 - isfp_inter[subframe], LP_ORDER);
1041
1042 2052 extrapolate_isf(e_isf);
1043
1044 2052 e_isf[LP_ORDER_16k - 1] *= 2.0;
1045 2052 ff_acelp_lsf2lspd(e_isp, e_isf, LP_ORDER_16k);
1046 2052 ff_amrwb_lsp2lpc(e_isp, hb_lpc, LP_ORDER_16k);
1047
1048 2052 lpc_weighting(hb_lpc, hb_lpc, 0.9, LP_ORDER_16k);
1049 } else {
1050 23020 lpc_weighting(hb_lpc, ctx->lp_coef[subframe], 0.6, LP_ORDER);
1051 }
1052
1053
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25072 ctx->celpf_ctx.celp_lp_synthesis_filterf(samples, hb_lpc, exc, AMRWB_SFR_SIZE_16k,
1054 (mode == MODE_6k60) ? LP_ORDER_16k : LP_ORDER);
1055 25072 }
1056
1057 /**
1058 * Apply a 15th order filter to high-band samples.
1059 * The filter characteristic depends on the given coefficients.
1060 *
1061 * @param[out] out Buffer for filtered output
1062 * @param[in] fir_coef Filter coefficients
1063 * @param[in,out] mem State from last filtering (updated)
1064 * @param[in] in Input speech data (high-band)
1065 *
1066 * @remark It is safe to pass the same array in in and out parameters
1067 */
1068
1069 #ifndef hb_fir_filter
1070 33728 static void hb_fir_filter(float *out, const float fir_coef[HB_FIR_SIZE + 1],
1071 float mem[HB_FIR_SIZE], const float *in)
1072 {
1073 int i, j;
1074 float data[AMRWB_SFR_SIZE_16k + HB_FIR_SIZE]; // past and current samples
1075
1076 33728 memcpy(data, mem, HB_FIR_SIZE * sizeof(float));
1077 33728 memcpy(data + HB_FIR_SIZE, in, AMRWB_SFR_SIZE_16k * sizeof(float));
1078
1079
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2731968 for (i = 0; i < AMRWB_SFR_SIZE_16k; i++) {
1080 2698240 out[i] = 0.0;
1081
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86343680 for (j = 0; j <= HB_FIR_SIZE; j++)
1082 83645440 out[i] += data[i + j] * fir_coef[j];
1083 }
1084
1085 33728 memcpy(mem, data + AMRWB_SFR_SIZE_16k, HB_FIR_SIZE * sizeof(float));
1086 33728 }
1087 #endif /* hb_fir_filter */
1088
1089 /**
1090 * Update context state before the next subframe.
1091 */
1092 25072 static void update_sub_state(AMRWBContext *ctx)
1093 {
1094 25072 memmove(&ctx->excitation_buf[0], &ctx->excitation_buf[AMRWB_SFR_SIZE],
1095 (AMRWB_P_DELAY_MAX + LP_ORDER + 1) * sizeof(float));
1096
1097 25072 memmove(&ctx->pitch_gain[1], &ctx->pitch_gain[0], 5 * sizeof(float));
1098 25072 memmove(&ctx->fixed_gain[1], &ctx->fixed_gain[0], 1 * sizeof(float));
1099
1100 25072 memmove(&ctx->samples_az[0], &ctx->samples_az[AMRWB_SFR_SIZE],
1101 LP_ORDER * sizeof(float));
1102 25072 memmove(&ctx->samples_up[0], &ctx->samples_up[AMRWB_SFR_SIZE],
1103 UPS_MEM_SIZE * sizeof(float));
1104 25072 memmove(&ctx->samples_hb[0], &ctx->samples_hb[AMRWB_SFR_SIZE_16k],
1105 LP_ORDER_16k * sizeof(float));
1106 25072 }
1107
1108 6268 static int amrwb_decode_frame(AVCodecContext *avctx, AVFrame *frame,
1109 int *got_frame_ptr, AVPacket *avpkt)
1110 {
1111 6268 AMRWBChannelsContext *s = avctx->priv_data;
1112 6268 const uint8_t *buf = avpkt->data;
1113 6268 int buf_size = avpkt->size;
1114 int sub, i, ret;
1115
1116 /* get output buffer */
1117 6268 frame->nb_samples = 4 * AMRWB_SFR_SIZE_16k;
1118
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6268 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
1119 return ret;
1120
1121
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12536 for (int ch = 0; ch < avctx->ch_layout.nb_channels; ch++) {
1122 6268 AMRWBContext *ctx = &s->ch[ch];
1123 6268 AMRWBFrame *cf = &ctx->frame;
1124 int expected_fr_size, header_size;
1125 float spare_vector[AMRWB_SFR_SIZE]; // extra stack space to hold result from anti-sparseness processing
1126 float fixed_gain_factor; // fixed gain correction factor (gamma)
1127 float *synth_fixed_vector; // pointer to the fixed vector that synthesis should use
1128 float synth_fixed_gain; // the fixed gain that synthesis should use
1129 float voice_fac, stab_fac; // parameters used for gain smoothing
1130 float synth_exc[AMRWB_SFR_SIZE]; // post-processed excitation for synthesis
1131 float hb_exc[AMRWB_SFR_SIZE_16k]; // excitation for the high frequency band
1132 float hb_samples[AMRWB_SFR_SIZE_16k]; // filtered high-band samples from synthesis
1133 float hb_gain;
1134 6268 float *buf_out = (float *)frame->extended_data[ch];
1135
1136 6268 header_size = decode_mime_header(ctx, buf);
1137 6268 expected_fr_size = ((cf_sizes_wb[ctx->fr_cur_mode] + 7) >> 3) + 1;
1138
1139
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6268 if (!ctx->fr_quality)
1140 av_log(avctx, AV_LOG_ERROR, "Encountered a bad or corrupted frame\n");
1141
1142
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6268 if (ctx->fr_cur_mode == NO_DATA || !ctx->fr_quality) {
1143 /* The specification suggests a "random signal" and
1144 "a muting technique" to "gradually decrease the output level". */
1145 av_samples_set_silence(&frame->extended_data[ch], 0, frame->nb_samples, 1, AV_SAMPLE_FMT_FLT);
1146 buf += expected_fr_size;
1147 buf_size -= expected_fr_size;
1148 continue;
1149 }
1150
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6268 if (ctx->fr_cur_mode > MODE_SID) {
1151 av_log(avctx, AV_LOG_ERROR,
1152 "Invalid mode %d\n", ctx->fr_cur_mode);
1153 return AVERROR_INVALIDDATA;
1154 }
1155
1156
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6268 if (buf_size < expected_fr_size) {
1157 av_log(avctx, AV_LOG_ERROR,
1158 "Frame too small (%d bytes). Truncated file?\n", buf_size);
1159 *got_frame_ptr = 0;
1160 return AVERROR_INVALIDDATA;
1161 }
1162
1163
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6268 if (ctx->fr_cur_mode == MODE_SID) { /* Comfort noise frame */
1164 avpriv_request_sample(avctx, "SID mode");
1165 return AVERROR_PATCHWELCOME;
1166 }
1167
1168 6268 ff_amr_bit_reorder((uint16_t *) &ctx->frame, sizeof(AMRWBFrame),
1169 6268 buf + header_size, amr_bit_orderings_by_mode[ctx->fr_cur_mode]);
1170
1171 /* Decode the quantized ISF vector */
1172
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6268 if (ctx->fr_cur_mode == MODE_6k60) {
1173 513 decode_isf_indices_36b(cf->isp_id, ctx->isf_cur);
1174 } else {
1175 5755 decode_isf_indices_46b(cf->isp_id, ctx->isf_cur);
1176 }
1177
1178 6268 isf_add_mean_and_past(ctx->isf_cur, ctx->isf_q_past);
1179 6268 ff_set_min_dist_lsf(ctx->isf_cur, MIN_ISF_SPACING, LP_ORDER - 1);
1180
1181 6268 stab_fac = stability_factor(ctx->isf_cur, ctx->isf_past_final);
1182
1183 6268 ctx->isf_cur[LP_ORDER - 1] *= 2.0;
1184 6268 ff_acelp_lsf2lspd(ctx->isp[3], ctx->isf_cur, LP_ORDER);
1185
1186 /* Generate a ISP vector for each subframe */
1187
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6268 if (ctx->first_frame) {
1188 22 ctx->first_frame = 0;
1189 22 memcpy(ctx->isp_sub4_past, ctx->isp[3], LP_ORDER * sizeof(double));
1190 }
1191 6268 interpolate_isp(ctx->isp, ctx->isp_sub4_past);
1192
1193
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31340 for (sub = 0; sub < 4; sub++)
1194 25072 ff_amrwb_lsp2lpc(ctx->isp[sub], ctx->lp_coef[sub], LP_ORDER);
1195
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31340 for (sub = 0; sub < 4; sub++) {
1197 25072 const AMRWBSubFrame *cur_subframe = &cf->subframe[sub];
1198 25072 float *sub_buf = buf_out + sub * AMRWB_SFR_SIZE_16k;
1199
1200 /* Decode adaptive codebook (pitch vector) */
1201 25072 decode_pitch_vector(ctx, cur_subframe, sub);
1202 /* Decode innovative codebook (fixed vector) */
1203 25072 decode_fixed_vector(ctx->fixed_vector, cur_subframe->pul_ih,
1204 25072 cur_subframe->pul_il, ctx->fr_cur_mode);
1205
1206 25072 pitch_sharpening(ctx, ctx->fixed_vector);
1207
1208 25072 decode_gains(cur_subframe->vq_gain, ctx->fr_cur_mode,
1209 &fixed_gain_factor, &ctx->pitch_gain[0]);
1210
1211 25072 ctx->fixed_gain[0] =
1212 25072 ff_amr_set_fixed_gain(fixed_gain_factor,
1213 25072 ctx->celpm_ctx.dot_productf(ctx->fixed_vector,
1214 25072 ctx->fixed_vector,
1215 AMRWB_SFR_SIZE) /
1216 AMRWB_SFR_SIZE,
1217 25072 ctx->prediction_error,
1218 ENERGY_MEAN, energy_pred_fac);
1219
1220 /* Calculate voice factor and store tilt for next subframe */
1221 25072 voice_fac = voice_factor(ctx->pitch_vector, ctx->pitch_gain[0],
1222 25072 ctx->fixed_vector, ctx->fixed_gain[0],
1223 &ctx->celpm_ctx);
1224 25072 ctx->tilt_coef = voice_fac * 0.25 + 0.25;
1225
1226 /* Construct current excitation */
1227
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1629680 for (i = 0; i < AMRWB_SFR_SIZE; i++) {
1228 1604608 ctx->excitation[i] *= ctx->pitch_gain[0];
1229 1604608 ctx->excitation[i] += ctx->fixed_gain[0] * ctx->fixed_vector[i];
1230 1604608 ctx->excitation[i] = truncf(ctx->excitation[i]);
1231 }
1232
1233 /* Post-processing of excitation elements */
1234 25072 synth_fixed_gain = noise_enhancer(ctx->fixed_gain[0], &ctx->prev_tr_gain,
1235 voice_fac, stab_fac);
1236
1237 25072 synth_fixed_vector = anti_sparseness(ctx, ctx->fixed_vector,
1238 spare_vector);
1239
1240 25072 pitch_enhancer(synth_fixed_vector, voice_fac);
1241
1242 25072 synthesis(ctx, ctx->lp_coef[sub], synth_exc, synth_fixed_gain,
1243 synth_fixed_vector, &ctx->samples_az[LP_ORDER]);
1244
1245 /* Synthesis speech post-processing */
1246 25072 de_emphasis(&ctx->samples_up[UPS_MEM_SIZE],
1247 25072 &ctx->samples_az[LP_ORDER], PREEMPH_FAC, ctx->demph_mem);
1248
1249 25072 ctx->acelpf_ctx.acelp_apply_order_2_transfer_function(&ctx->samples_up[UPS_MEM_SIZE],
1250 25072 &ctx->samples_up[UPS_MEM_SIZE], hpf_zeros, hpf_31_poles,
1251 25072 hpf_31_gain, ctx->hpf_31_mem, AMRWB_SFR_SIZE);
1252
1253 25072 upsample_5_4(sub_buf, &ctx->samples_up[UPS_FIR_SIZE],
1254 AMRWB_SFR_SIZE_16k, &ctx->celpm_ctx);
1255
1256 /* High frequency band (6.4 - 7.0 kHz) generation part */
1257 25072 ctx->acelpf_ctx.acelp_apply_order_2_transfer_function(hb_samples,
1258 25072 &ctx->samples_up[UPS_MEM_SIZE], hpf_zeros, hpf_400_poles,
1259 25072 hpf_400_gain, ctx->hpf_400_mem, AMRWB_SFR_SIZE);
1260
1261 25072 hb_gain = find_hb_gain(ctx, hb_samples,
1262 25072 cur_subframe->hb_gain, cf->vad);
1263
1264 25072 scaled_hb_excitation(ctx, hb_exc, synth_exc, hb_gain);
1265
1266 25072 hb_synthesis(ctx, sub, &ctx->samples_hb[LP_ORDER_16k],
1267 25072 hb_exc, ctx->isf_cur, ctx->isf_past_final);
1268
1269 /* High-band post-processing filters */
1270 25072 hb_fir_filter(hb_samples, bpf_6_7_coef, ctx->bpf_6_7_mem,
1271 25072 &ctx->samples_hb[LP_ORDER_16k]);
1272
1273
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25072 if (ctx->fr_cur_mode == MODE_23k85)
1274 8656 hb_fir_filter(hb_samples, lpf_7_coef, ctx->lpf_7_mem,
1275 hb_samples);
1276
1277 /* Add the low and high frequency bands */
1278
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2030832 for (i = 0; i < AMRWB_SFR_SIZE_16k; i++)
1279 2005760 sub_buf[i] = (sub_buf[i] + hb_samples[i]) * (1.0f / (1 << 15));
1280
1281 /* Update buffers and history */
1282 25072 update_sub_state(ctx);
1283 }
1284
1285 /* update state for next frame */
1286 6268 memcpy(ctx->isp_sub4_past, ctx->isp[3], LP_ORDER * sizeof(ctx->isp[3][0]));
1287 6268 memcpy(ctx->isf_past_final, ctx->isf_cur, LP_ORDER * sizeof(float));
1288
1289 6268 buf += expected_fr_size;
1290 6268 buf_size -= expected_fr_size;
1291 }
1292
1293 6268 *got_frame_ptr = 1;
1294
1295 6268 return buf - avpkt->data;
1296 }
1297
1298 const FFCodec ff_amrwb_decoder = {
1299 .p.name = "amrwb",
1300 CODEC_LONG_NAME("AMR-WB (Adaptive Multi-Rate WideBand)"),
1301 .p.type = AVMEDIA_TYPE_AUDIO,
1302 .p.id = AV_CODEC_ID_AMR_WB,
1303 .priv_data_size = sizeof(AMRWBChannelsContext),
1304 .init = amrwb_decode_init,
1305 FF_CODEC_DECODE_CB(amrwb_decode_frame),
1306 .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF,
1307 .p.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP,
1308 AV_SAMPLE_FMT_NONE },
1309 };
1310