FFmpeg coverage

Directory: ../../../ffmpeg/
File: src/libavcodec/amrwbdec.c
Date: 2024-07-26 21:54:09
Exec Total Coverage
Lines: 498 520 95.8%
Functions: 35 35 100.0%
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/avassert.h"
30 #include "libavutil/channel_layout.h"
31 #include "libavutil/common.h"
32 #include "libavutil/lfg.h"
33
34 #include "avcodec.h"
35 #include "lsp.h"
36 #include "celp_filters.h"
37 #include "celp_math.h"
38 #include "acelp_filters.h"
39 #include "acelp_vectors.h"
40 #include "acelp_pitch_delay.h"
41 #include "codec_internal.h"
42 #include "decode.h"
43
44 #define AMR_USE_16BIT_TABLES
45 #include "amr.h"
46
47 #include "amrwbdata.h"
48 #if ARCH_MIPS
49 #include "mips/amrwbdec_mips.h"
50 #endif /* ARCH_MIPS */
51
52 typedef struct AMRWBContext {
53 AMRWBFrame frame; ///< AMRWB parameters decoded from bitstream
54 enum Mode fr_cur_mode; ///< mode index of current frame
55 uint8_t fr_quality; ///< frame quality index (FQI)
56 float isf_cur[LP_ORDER]; ///< working ISF vector from current frame
57 float isf_q_past[LP_ORDER]; ///< quantized ISF vector of the previous frame
58 float isf_past_final[LP_ORDER]; ///< final processed ISF vector of the previous frame
59 double isp[4][LP_ORDER]; ///< ISP vectors from current frame
60 double isp_sub4_past[LP_ORDER]; ///< ISP vector for the 4th subframe of the previous frame
61
62 float lp_coef[4][LP_ORDER]; ///< Linear Prediction Coefficients from ISP vector
63
64 uint8_t base_pitch_lag; ///< integer part of pitch lag for the next relative subframe
65 uint8_t pitch_lag_int; ///< integer part of pitch lag of the previous subframe
66
67 float excitation_buf[AMRWB_P_DELAY_MAX + LP_ORDER + 2 + AMRWB_SFR_SIZE]; ///< current excitation and all necessary excitation history
68 float *excitation; ///< points to current excitation in excitation_buf[]
69
70 float pitch_vector[AMRWB_SFR_SIZE]; ///< adaptive codebook (pitch) vector for current subframe
71 float fixed_vector[AMRWB_SFR_SIZE]; ///< algebraic codebook (fixed) vector for current subframe
72
73 float prediction_error[4]; ///< quantified prediction errors {20log10(^gamma_gc)} for previous four subframes
74 float pitch_gain[6]; ///< quantified pitch gains for the current and previous five subframes
75 float fixed_gain[2]; ///< quantified fixed gains for the current and previous subframes
76
77 float tilt_coef; ///< {beta_1} related to the voicing of the previous subframe
78
79 float prev_sparse_fixed_gain; ///< previous fixed gain; used by anti-sparseness to determine "onset"
80 uint8_t prev_ir_filter_nr; ///< previous impulse response filter "impNr": 0 - strong, 1 - medium, 2 - none
81 float prev_tr_gain; ///< previous initial gain used by noise enhancer for threshold
82
83 float samples_az[LP_ORDER + AMRWB_SFR_SIZE]; ///< low-band samples and memory from synthesis at 12.8kHz
84 float samples_up[UPS_MEM_SIZE + AMRWB_SFR_SIZE]; ///< low-band samples and memory processed for upsampling
85 float samples_hb[LP_ORDER_16k + AMRWB_SFR_SIZE_16k]; ///< high-band samples and memory from synthesis at 16kHz
86
87 float hpf_31_mem[2], hpf_400_mem[2]; ///< previous values in the high pass filters
88 float demph_mem[1]; ///< previous value in the de-emphasis filter
89 float bpf_6_7_mem[HB_FIR_SIZE]; ///< previous values in the high-band band pass filter
90 float lpf_7_mem[HB_FIR_SIZE]; ///< previous values in the high-band low pass filter
91
92 AVLFG prng; ///< random number generator for white noise excitation
93 uint8_t first_frame; ///< flag active during decoding of the first frame
94 ACELPFContext acelpf_ctx; ///< context for filters for ACELP-based codecs
95 ACELPVContext acelpv_ctx; ///< context for vector operations for ACELP-based codecs
96 CELPFContext celpf_ctx; ///< context for filters for CELP-based codecs
97 CELPMContext celpm_ctx; ///< context for fixed point math operations
98
99 } AMRWBContext;
100
101 typedef struct AMRWBChannelsContext {
102 AMRWBContext ch[2];
103 } AMRWBChannelsContext;
104
105 22 static av_cold int amrwb_decode_init(AVCodecContext *avctx)
106 {
107 22 AMRWBChannelsContext *s = avctx->priv_data;
108 int i;
109
110
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 22 if (avctx->ch_layout.nb_channels > 2) {
111 ✗ avpriv_report_missing_feature(avctx, ">2 channel AMR");
112 ✗ return AVERROR_PATCHWELCOME;
113 }
114
115
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 22 if (!avctx->ch_layout.nb_channels) {
116 ✗ av_channel_layout_uninit(&avctx->ch_layout);
117 ✗ avctx->ch_layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_MONO;
118 }
119
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 22 if (!avctx->sample_rate)
120 ✗ avctx->sample_rate = 16000;
121 22 avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
122
123
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 44 for (int ch = 0; ch < avctx->ch_layout.nb_channels; ch++) {
124 22 AMRWBContext *ctx = &s->ch[ch];
125
126 22 av_lfg_init(&ctx->prng, 1);
127
128 22 ctx->excitation = &ctx->excitation_buf[AMRWB_P_DELAY_MAX + LP_ORDER + 1];
129 22 ctx->first_frame = 1;
130
131
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 374 for (i = 0; i < LP_ORDER; i++)
132 352 ctx->isf_past_final[i] = isf_init[i] * (1.0f / (1 << 15));
133
134
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 110 for (i = 0; i < 4; i++)
135 88 ctx->prediction_error[i] = MIN_ENERGY;
136
137 22 ff_acelp_filter_init(&ctx->acelpf_ctx);
138 22 ff_acelp_vectors_init(&ctx->acelpv_ctx);
139 22 ff_celp_filter_init(&ctx->celpf_ctx);
140 22 ff_celp_math_init(&ctx->celpm_ctx);
141 }
142
143 22 return 0;
144 }
145
146 /**
147 * Decode the frame header in the "MIME/storage" format. This format
148 * is simpler and does not carry the auxiliary frame information.
149 *
150 * @param[in] ctx The Context
151 * @param[in] buf Pointer to the input buffer
152 *
153 * @return The decoded header length in bytes
154 */
155 6268 static int decode_mime_header(AMRWBContext *ctx, const uint8_t *buf)
156 {
157 /* Decode frame header (1st octet) */
158 6268 ctx->fr_cur_mode = buf[0] >> 3 & 0x0F;
159 6268 ctx->fr_quality = (buf[0] & 0x4) == 0x4;
160
161 6268 return 1;
162 }
163
164 /**
165 * Decode quantized ISF vectors using 36-bit indexes (6K60 mode only).
166 *
167 * @param[in] ind Array of 5 indexes
168 * @param[out] isf_q Buffer for isf_q[LP_ORDER]
169 */
170 513 static void decode_isf_indices_36b(uint16_t *ind, float *isf_q)
171 {
172 int i;
173
174
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 5130 for (i = 0; i < 9; i++)
175 4617 isf_q[i] = dico1_isf[ind[0]][i] * (1.0f / (1 << 15));
176
177
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 4104 for (i = 0; i < 7; i++)
178 3591 isf_q[i + 9] = dico2_isf[ind[1]][i] * (1.0f / (1 << 15));
179
180
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 3078 for (i = 0; i < 5; i++)
181 2565 isf_q[i] += dico21_isf_36b[ind[2]][i] * (1.0f / (1 << 15));
182
183
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 2565 for (i = 0; i < 4; i++)
184 2052 isf_q[i + 5] += dico22_isf_36b[ind[3]][i] * (1.0f / (1 << 15));
185
186
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 4104 for (i = 0; i < 7; i++)
187 3591 isf_q[i + 9] += dico23_isf_36b[ind[4]][i] * (1.0f / (1 << 15));
188 513 }
189
190 /**
191 * Decode quantized ISF vectors using 46-bit indexes (except 6K60 mode).
192 *
193 * @param[in] ind Array of 7 indexes
194 * @param[out] isf_q Buffer for isf_q[LP_ORDER]
195 */
196 5755 static void decode_isf_indices_46b(uint16_t *ind, float *isf_q)
197 {
198 int i;
199
200
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201 51795 isf_q[i] = dico1_isf[ind[0]][i] * (1.0f / (1 << 15));
202
203
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 46040 for (i = 0; i < 7; i++)
204 40285 isf_q[i + 9] = dico2_isf[ind[1]][i] * (1.0f / (1 << 15));
205
206
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 23020 for (i = 0; i < 3; i++)
207 17265 isf_q[i] += dico21_isf[ind[2]][i] * (1.0f / (1 << 15));
208
209
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 23020 for (i = 0; i < 3; i++)
210 17265 isf_q[i + 3] += dico22_isf[ind[3]][i] * (1.0f / (1 << 15));
211
212
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 23020 for (i = 0; i < 3; i++)
213 17265 isf_q[i + 6] += dico23_isf[ind[4]][i] * (1.0f / (1 << 15));
214
215
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 23020 for (i = 0; i < 3; i++)
216 17265 isf_q[i + 9] += dico24_isf[ind[5]][i] * (1.0f / (1 << 15));
217
218
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 28775 for (i = 0; i < 4; i++)
219 23020 isf_q[i + 12] += dico25_isf[ind[6]][i] * (1.0f / (1 << 15));
220 5755 }
221
222 /**
223 * Apply mean and past ISF values using the prediction factor.
224 * Updates past ISF vector.
225 *
226 * @param[in,out] isf_q Current quantized ISF
227 * @param[in,out] isf_past Past quantized ISF
228 */
229 6268 static void isf_add_mean_and_past(float *isf_q, float *isf_past)
230 {
231 int i;
232 float tmp;
233
234
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 106556 for (i = 0; i < LP_ORDER; i++) {
235 100288 tmp = isf_q[i];
236 100288 isf_q[i] += isf_mean[i] * (1.0f / (1 << 15));
237 100288 isf_q[i] += PRED_FACTOR * isf_past[i];
238 100288 isf_past[i] = tmp;
239 }
240 6268 }
241
242 /**
243 * Interpolate the fourth ISP vector from current and past frames
244 * to obtain an ISP vector for each subframe.
245 *
246 * @param[in,out] isp_q ISPs for each subframe
247 * @param[in] isp4_past Past ISP for subframe 4
248 */
249 6268 static void interpolate_isp(double isp_q[4][LP_ORDER], const double *isp4_past)
250 {
251 int i, k;
252
253
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 25072 for (k = 0; k < 3; k++) {
254 18804 float c = isfp_inter[k];
255
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 319668 for (i = 0; i < LP_ORDER; i++)
256 300864 isp_q[k][i] = (1.0 - c) * isp4_past[i] + c * isp_q[3][i];
257 }
258 6268 }
259
260 /**
261 * Decode an adaptive codebook index into pitch lag (except 6k60, 8k85 modes).
262 * Calculate integer lag and fractional lag always using 1/4 resolution.
263 * In 1st and 3rd subframes the index is relative to last subframe integer lag.
264 *
265 * @param[out] lag_int Decoded integer pitch lag
266 * @param[out] lag_frac Decoded fractional pitch lag
267 * @param[in] pitch_index Adaptive codebook pitch index
268 * @param[in,out] base_lag_int Base integer lag used in relative subframes
269 * @param[in] subframe Current subframe index (0 to 3)
270 */
271 20968 static void decode_pitch_lag_high(int *lag_int, int *lag_frac, int pitch_index,
272 uint8_t *base_lag_int, int subframe)
273 {
274
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 20968 if (subframe == 0 || subframe == 2) {
275
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 10484 if (pitch_index < 376) {
276 7630 *lag_int = (pitch_index + 137) >> 2;
277 7630 *lag_frac = pitch_index - (*lag_int << 2) + 136;
278
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 2854 } else if (pitch_index < 440) {
279 1248 *lag_int = (pitch_index + 257 - 376) >> 1;
280 1248 *lag_frac = (pitch_index - (*lag_int << 1) + 256 - 376) * 2;
281 /* the actual resolution is 1/2 but expressed as 1/4 */
282 } else {
283 1606 *lag_int = pitch_index - 280;
284 1606 *lag_frac = 0;
285 }
286 /* minimum lag for next subframe */
287 10484 *base_lag_int = av_clip(*lag_int - 8 - (*lag_frac < 0),
288 AMRWB_P_DELAY_MIN, AMRWB_P_DELAY_MAX - 15);
289 // XXX: the spec states clearly that *base_lag_int should be
290 // the nearest integer to *lag_int (minus 8), but the ref code
291 // actually always uses its floor, I'm following the latter
292 } else {
293 10484 *lag_int = (pitch_index + 1) >> 2;
294 10484 *lag_frac = pitch_index - (*lag_int << 2);
295 10484 *lag_int += *base_lag_int;
296 }
297 20968 }
298
299 /**
300 * Decode an adaptive codebook index into pitch lag for 8k85 and 6k60 modes.
301 * The description is analogous to decode_pitch_lag_high, but in 6k60 the
302 * relative index is used for all subframes except the first.
303 */
304 4104 static void decode_pitch_lag_low(int *lag_int, int *lag_frac, int pitch_index,
305 uint8_t *base_lag_int, int subframe, enum Mode mode)
306 {
307
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 4104 if (subframe == 0 || (subframe == 2 && mode != MODE_6k60)) {
308
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 1539 if (pitch_index < 116) {
309 850 *lag_int = (pitch_index + 69) >> 1;
310 850 *lag_frac = (pitch_index - (*lag_int << 1) + 68) * 2;
311 } else {
312 689 *lag_int = pitch_index - 24;
313 689 *lag_frac = 0;
314 }
315 // XXX: same problem as before
316 1539 *base_lag_int = av_clip(*lag_int - 8 - (*lag_frac < 0),
317 AMRWB_P_DELAY_MIN, AMRWB_P_DELAY_MAX - 15);
318 } else {
319 2565 *lag_int = (pitch_index + 1) >> 1;
320 2565 *lag_frac = (pitch_index - (*lag_int << 1)) * 2;
321 2565 *lag_int += *base_lag_int;
322 }
323 4104 }
324
325 /**
326 * Find the pitch vector by interpolating the past excitation at the
327 * pitch delay, which is obtained in this function.
328 *
329 * @param[in,out] ctx The context
330 * @param[in] amr_subframe Current subframe data
331 * @param[in] subframe Current subframe index (0 to 3)
332 */
333 25072 static void decode_pitch_vector(AMRWBContext *ctx,
334 const AMRWBSubFrame *amr_subframe,
335 const int subframe)
336 {
337 int pitch_lag_int, pitch_lag_frac;
338 int i;
339 25072 float *exc = ctx->excitation;
340 25072 enum Mode mode = ctx->fr_cur_mode;
341
342
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 25072 if (mode <= MODE_8k85) {
343 4104 decode_pitch_lag_low(&pitch_lag_int, &pitch_lag_frac, amr_subframe->adap,
344 &ctx->base_pitch_lag, subframe, mode);
345 } else
346 20968 decode_pitch_lag_high(&pitch_lag_int, &pitch_lag_frac, amr_subframe->adap,
347 &ctx->base_pitch_lag, subframe);
348
349 25072 ctx->pitch_lag_int = pitch_lag_int;
350 25072 pitch_lag_int += pitch_lag_frac > 0;
351
352 /* Calculate the pitch vector by interpolating the past excitation at the
353 pitch lag using a hamming windowed sinc function */
354 50144 ctx->acelpf_ctx.acelp_interpolatef(exc,
355 25072 exc + 1 - pitch_lag_int,
356 ac_inter, 4,
357
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 25072 pitch_lag_frac + (pitch_lag_frac > 0 ? 0 : 4),
358 LP_ORDER, AMRWB_SFR_SIZE + 1);
359
360 /* Check which pitch signal path should be used
361 * 6k60 and 8k85 modes have the ltp flag set to 0 */
362
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 25072 if (amr_subframe->ltp) {
363 7140 memcpy(ctx->pitch_vector, exc, AMRWB_SFR_SIZE * sizeof(float));
364 } else {
365
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 1165580 for (i = 0; i < AMRWB_SFR_SIZE; i++)
366 1147648 ctx->pitch_vector[i] = 0.18 * exc[i - 1] + 0.64 * exc[i] +
367 1147648 0.18 * exc[i + 1];
368 17932 memcpy(exc, ctx->pitch_vector, AMRWB_SFR_SIZE * sizeof(float));
369 }
370 25072 }
371
372 /** Get x bits in the index interval [lsb,lsb+len-1] inclusive */
373 #define BIT_STR(x,lsb,len) av_zero_extend((x) >> (lsb), (len))
374
375 /** Get the bit at specified position */
376 #define BIT_POS(x, p) (((x) >> (p)) & 1)
377
378 /**
379 * The next six functions decode_[i]p_track decode exactly i pulses
380 * positions and amplitudes (-1 or 1) in a subframe track using
381 * an encoded pulse indexing (TS 26.190 section 5.8.2).
382 *
383 * The results are given in out[], in which a negative number means
384 * amplitude -1 and vice versa (i.e., ampl(x) = x / abs(x) ).
385 *
386 * @param[out] out Output buffer (writes i elements)
387 * @param[in] code Pulse index (no. of bits varies, see below)
388 * @param[in] m (log2) Number of potential positions
389 * @param[in] off Offset for decoded positions
390 */
391 106180 static inline void decode_1p_track(int *out, int code, int m, int off)
392 {
393 106180 int pos = BIT_STR(code, 0, m) + off; ///code: m+1 bits
394
395
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 106180 out[0] = BIT_POS(code, m) ? -pos : pos;
396 106180 }
397
398 147226 static inline void decode_2p_track(int *out, int code, int m, int off) ///code: 2m+1 bits
399 {
400 147226 int pos0 = BIT_STR(code, m, m) + off;
401 147226 int pos1 = BIT_STR(code, 0, m) + off;
402
403
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 147226 out[0] = BIT_POS(code, 2*m) ? -pos0 : pos0;
404
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 147226 out[1] = BIT_POS(code, 2*m) ? -pos1 : pos1;
405
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 147226 out[1] = pos0 > pos1 ? -out[1] : out[1];
406 147226 }
407
408 68748 static void decode_3p_track(int *out, int code, int m, int off) ///code: 3m+1 bits
409 {
410 68748 int half_2p = BIT_POS(code, 2*m - 1) << (m - 1);
411
412 68748 decode_2p_track(out, BIT_STR(code, 0, 2*m - 1),
413 m - 1, off + half_2p);
414 68748 decode_1p_track(out + 2, BIT_STR(code, 2*m, m + 1), m, off);
415 68748 }
416
417 32563 static void decode_4p_track(int *out, int code, int m, int off) ///code: 4m bits
418 {
419 int half_4p, subhalf_2p;
420 32563 int b_offset = 1 << (m - 1);
421
422
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 32563 switch (BIT_STR(code, 4*m - 2, 2)) { /* case ID (2 bits) */
423 3612 case 0: /* 0 pulses in A, 4 pulses in B or vice versa */
424 3612 half_4p = BIT_POS(code, 4*m - 3) << (m - 1); // which has 4 pulses
425 3612 subhalf_2p = BIT_POS(code, 2*m - 3) << (m - 2);
426
427 3612 decode_2p_track(out, BIT_STR(code, 0, 2*m - 3),
428 3612 m - 2, off + half_4p + subhalf_2p);
429 3612 decode_2p_track(out + 2, BIT_STR(code, 2*m - 2, 2*m - 1),
430 m - 1, off + half_4p);
431 3612 break;
432 7991 case 1: /* 1 pulse in A, 3 pulses in B */
433 7991 decode_1p_track(out, BIT_STR(code, 3*m - 2, m),
434 m - 1, off);
435 7991 decode_3p_track(out + 1, BIT_STR(code, 0, 3*m - 2),
436 m - 1, off + b_offset);
437 7991 break;
438 12806 case 2: /* 2 pulses in each half */
439 12806 decode_2p_track(out, BIT_STR(code, 2*m - 1, 2*m - 1),
440 m - 1, off);
441 12806 decode_2p_track(out + 2, BIT_STR(code, 0, 2*m - 1),
442 m - 1, off + b_offset);
443 12806 break;
444 8154 case 3: /* 3 pulses in A, 1 pulse in B */
445 8154 decode_3p_track(out, BIT_STR(code, m, 3*m - 2),
446 m - 1, off);
447 8154 decode_1p_track(out + 3, BIT_STR(code, 0, m),
448 m - 1, off + b_offset);
449 8154 break;
450 }
451 32563 }
452
453 13079 static void decode_5p_track(int *out, int code, int m, int off) ///code: 5m bits
454 {
455 13079 int half_3p = BIT_POS(code, 5*m - 1) << (m - 1);
456
457 13079 decode_3p_track(out, BIT_STR(code, 2*m + 1, 3*m - 2),
458 m - 1, off + half_3p);
459
460 13079 decode_2p_track(out + 3, BIT_STR(code, 0, 2*m + 1), m, off);
461 13079 }
462
463 42832 static void decode_6p_track(int *out, int code, int m, int off) ///code: 6m-2 bits
464 {
465 42832 int b_offset = 1 << (m - 1);
466 /* which half has more pulses in cases 0 to 2 */
467 42832 int half_more = BIT_POS(code, 6*m - 5) << (m - 1);
468 42832 int half_other = b_offset - half_more;
469
470
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 42832 switch (BIT_STR(code, 6*m - 4, 2)) { /* case ID (2 bits) */
471 1297 case 0: /* 0 pulses in A, 6 pulses in B or vice versa */
472 1297 decode_1p_track(out, BIT_STR(code, 0, m),
473 m - 1, off + half_more);
474 1297 decode_5p_track(out + 1, BIT_STR(code, m, 5*m - 5),
475 m - 1, off + half_more);
476 1297 break;
477 7678 case 1: /* 1 pulse in A, 5 pulses in B or vice versa */
478 7678 decode_1p_track(out, BIT_STR(code, 0, m),
479 m - 1, off + half_other);
480 7678 decode_5p_track(out + 1, BIT_STR(code, m, 5*m - 5),
481 m - 1, off + half_more);
482 7678 break;
483 20251 case 2: /* 2 pulses in A, 4 pulses in B or vice versa */
484 20251 decode_2p_track(out, BIT_STR(code, 0, 2*m - 1),
485 m - 1, off + half_other);
486 20251 decode_4p_track(out + 2, BIT_STR(code, 2*m - 1, 4*m - 4),
487 m - 1, off + half_more);
488 20251 break;
489 13606 case 3: /* 3 pulses in A, 3 pulses in B */
490 13606 decode_3p_track(out, BIT_STR(code, 3*m - 2, 3*m - 2),
491 m - 1, off);
492 13606 decode_3p_track(out + 3, BIT_STR(code, 0, 3*m - 2),
493 m - 1, off + b_offset);
494 13606 break;
495 }
496 42832 }
497
498 /**
499 * Decode the algebraic codebook index to pulse positions and signs,
500 * then construct the algebraic codebook vector.
501 *
502 * @param[out] fixed_vector Buffer for the fixed codebook excitation
503 * @param[in] pulse_hi MSBs part of the pulse index array (higher modes only)
504 * @param[in] pulse_lo LSBs part of the pulse index array
505 * @param[in] mode Mode of the current frame
506 */
507 25072 static void decode_fixed_vector(float *fixed_vector, const uint16_t *pulse_hi,
508 const uint16_t *pulse_lo, const enum Mode mode)
509 {
510 /* sig_pos stores for each track the decoded pulse position indexes
511 * (1-based) multiplied by its corresponding amplitude (+1 or -1) */
512 int sig_pos[4][6];
513
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 25072 int spacing = (mode == MODE_6k60) ? 2 : 4;
514 int i, j;
515
516
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 25072 switch (mode) {
517 2052 case MODE_6k60:
518
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 6156 for (i = 0; i < 2; i++)
519 4104 decode_1p_track(sig_pos[i], pulse_lo[i], 5, 1);
520 2052 break;
521 2052 case MODE_8k85:
522
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 10260 for (i = 0; i < 4; i++)
523 8208 decode_1p_track(sig_pos[i], pulse_lo[i], 4, 1);
524 2052 break;
525 2052 case MODE_12k65:
526
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 10260 for (i = 0; i < 4; i++)
527 8208 decode_2p_track(sig_pos[i], pulse_lo[i], 4, 1);
528 2052 break;
529 2052 case MODE_14k25:
530
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 6156 for (i = 0; i < 2; i++)
531 4104 decode_3p_track(sig_pos[i], pulse_lo[i], 4, 1);
532
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 6156 for (i = 2; i < 4; i++)
533 4104 decode_2p_track(sig_pos[i], pulse_lo[i], 4, 1);
534 2052 break;
535 2052 case MODE_15k85:
536
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 10260 for (i = 0; i < 4; i++)
537 8208 decode_3p_track(sig_pos[i], pulse_lo[i], 4, 1);
538 2052 break;
539 2052 case MODE_18k25:
540
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 10260 for (i = 0; i < 4; i++)
541 8208 decode_4p_track(sig_pos[i], (int) pulse_lo[i] +
542 8208 ((int) pulse_hi[i] << 14), 4, 1);
543 2052 break;
544 2052 case MODE_19k85:
545
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 6156 for (i = 0; i < 2; i++)
546 4104 decode_5p_track(sig_pos[i], (int) pulse_lo[i] +
547 4104 ((int) pulse_hi[i] << 10), 4, 1);
548
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 6156 for (i = 2; i < 4; i++)
549 4104 decode_4p_track(sig_pos[i], (int) pulse_lo[i] +
550 4104 ((int) pulse_hi[i] << 14), 4, 1);
551 2052 break;
552 10708 case MODE_23k05:
553 case MODE_23k85:
554
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 53540 for (i = 0; i < 4; i++)
555 42832 decode_6p_track(sig_pos[i], (int) pulse_lo[i] +
556 42832 ((int) pulse_hi[i] << 11), 4, 1);
557 10708 break;
558 25072 default:
559 av_assert2(0);
560 }
561
562 25072 memset(fixed_vector, 0, sizeof(float) * AMRWB_SFR_SIZE);
563
564
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 125360 for (i = 0; i < 4; i++)
565
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 500920 for (j = 0; j < pulses_nb_per_mode_tr[mode][i]; j++) {
566 400632 int pos = (FFABS(sig_pos[i][j]) - 1) * spacing + i;
567
568
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 400632 fixed_vector[pos] += sig_pos[i][j] < 0 ? -1.0 : 1.0;
569 }
570 25072 }
571
572 /**
573 * Decode pitch gain and fixed gain correction factor.
574 *
575 * @param[in] vq_gain Vector-quantized index for gains
576 * @param[in] mode Mode of the current frame
577 * @param[out] fixed_gain_factor Decoded fixed gain correction factor
578 * @param[out] pitch_gain Decoded pitch gain
579 */
580 25072 static void decode_gains(const uint8_t vq_gain, const enum Mode mode,
581 float *fixed_gain_factor, float *pitch_gain)
582 {
583
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 25072 const int16_t *gains = (mode <= MODE_8k85 ? qua_gain_6b[vq_gain] :
584 20968 qua_gain_7b[vq_gain]);
585
586 25072 *pitch_gain = gains[0] * (1.0f / (1 << 14));
587 25072 *fixed_gain_factor = gains[1] * (1.0f / (1 << 11));
588 25072 }
589
590 /**
591 * Apply pitch sharpening filters to the fixed codebook vector.
592 *
593 * @param[in] ctx The context
594 * @param[in,out] fixed_vector Fixed codebook excitation
595 */
596 // XXX: Spec states this procedure should be applied when the pitch
597 // lag is less than 64, but this checking seems absent in reference and AMR-NB
598 25072 static void pitch_sharpening(AMRWBContext *ctx, float *fixed_vector)
599 {
600 int i;
601
602 /* Tilt part */
603
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 1604608 for (i = AMRWB_SFR_SIZE - 1; i != 0; i--)
604 1579536 fixed_vector[i] -= fixed_vector[i - 1] * ctx->tilt_coef;
605
606 /* Periodicity enhancement part */
607
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 187025 for (i = ctx->pitch_lag_int; i < AMRWB_SFR_SIZE; i++)
608 161953 fixed_vector[i] += fixed_vector[i - ctx->pitch_lag_int] * 0.85;
609 25072 }
610
611 /**
612 * Calculate the voicing factor (-1.0 = unvoiced to 1.0 = voiced).
613 *
614 * @param[in] p_vector, f_vector Pitch and fixed excitation vectors
615 * @param[in] p_gain, f_gain Pitch and fixed gains
616 * @param[in] ctx The context
617 */
618 // XXX: There is something wrong with the precision here! The magnitudes
619 // of the energies are not correct. Please check the reference code carefully
620 25072 static float voice_factor(float *p_vector, float p_gain,
621 float *f_vector, float f_gain,
622 CELPMContext *ctx)
623 {
624 25072 double p_ener = (double) ctx->dot_productf(p_vector, p_vector,
625 25072 AMRWB_SFR_SIZE) *
626 25072 p_gain * p_gain;
627 25072 double f_ener = (double) ctx->dot_productf(f_vector, f_vector,
628 25072 AMRWB_SFR_SIZE) *
629 25072 f_gain * f_gain;
630
631 25072 return (p_ener - f_ener) / (p_ener + f_ener + 0.01);
632 }
633
634 /**
635 * Reduce fixed vector sparseness by smoothing with one of three IR filters,
636 * also known as "adaptive phase dispersion".
637 *
638 * @param[in] ctx The context
639 * @param[in,out] fixed_vector Unfiltered fixed vector
640 * @param[out] buf Space for modified vector if necessary
641 *
642 * @return The potentially overwritten filtered fixed vector address
643 */
644 25072 static float *anti_sparseness(AMRWBContext *ctx,
645 float *fixed_vector, float *buf)
646 {
647 int ir_filter_nr;
648
649
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 25072 if (ctx->fr_cur_mode > MODE_8k85) // no filtering in higher modes
650 20968 return fixed_vector;
651
652
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 4104 if (ctx->pitch_gain[0] < 0.6) {
653 2261 ir_filter_nr = 0; // strong filtering
654
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 1843 } else if (ctx->pitch_gain[0] < 0.9) {
655 712 ir_filter_nr = 1; // medium filtering
656 } else
657 1131 ir_filter_nr = 2; // no filtering
658
659 /* detect 'onset' */
660
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 4104 if (ctx->fixed_gain[0] > 3.0 * ctx->fixed_gain[1]) {
661
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 61 if (ir_filter_nr < 2)
662 39 ir_filter_nr++;
663 } else {
664 4043 int i, count = 0;
665
666
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667
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 24258 if (ctx->pitch_gain[i] < 0.6)
668 13373 count++;
669
670
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 4043 if (count > 2)
671 2724 ir_filter_nr = 0;
672
673
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 4043 if (ir_filter_nr > ctx->prev_ir_filter_nr + 1)
674 83 ir_filter_nr--;
675 }
676
677 /* update ir filter strength history */
678 4104 ctx->prev_ir_filter_nr = ir_filter_nr;
679
680 4104 ir_filter_nr += (ctx->fr_cur_mode == MODE_8k85);
681
682
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 4104 if (ir_filter_nr < 2) {
683 int i;
684 3197 const float *coef = ir_filters_lookup[ir_filter_nr];
685
686 /* Circular convolution code in the reference
687 * decoder was modified to avoid using one
688 * extra array. The filtered vector is given by:
689 *
690 * c2(n) = sum(i,0,len-1){ c(i) * coef( (n - i + len) % len ) }
691 */
692
693 3197 memset(buf, 0, sizeof(float) * AMRWB_SFR_SIZE);
694
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 207805 for (i = 0; i < AMRWB_SFR_SIZE; i++)
695
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 204608 if (fixed_vector[i])
696 21471 ff_celp_circ_addf(buf, buf, coef, i, fixed_vector[i],
697 AMRWB_SFR_SIZE);
698 3197 fixed_vector = buf;
699 }
700
701 4104 return fixed_vector;
702 }
703
704 /**
705 * Calculate a stability factor {teta} based on distance between
706 * current and past isf. A value of 1 shows maximum signal stability.
707 */
708 6268 static float stability_factor(const float *isf, const float *isf_past)
709 {
710 int i;
711 6268 float acc = 0.0;
712
713
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 100288 for (i = 0; i < LP_ORDER - 1; i++)
714 94020 acc += (isf[i] - isf_past[i]) * (isf[i] - isf_past[i]);
715
716 // XXX: This part is not so clear from the reference code
717 // the result is more accurate changing the "/ 256" to "* 512"
718
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 6268 return FFMAX(0.0, 1.25 - acc * 0.8 * 512);
719 }
720
721 /**
722 * Apply a non-linear fixed gain smoothing in order to reduce
723 * fluctuation in the energy of excitation.
724 *
725 * @param[in] fixed_gain Unsmoothed fixed gain
726 * @param[in,out] prev_tr_gain Previous threshold gain (updated)
727 * @param[in] voice_fac Frame voicing factor
728 * @param[in] stab_fac Frame stability factor
729 *
730 * @return The smoothed gain
731 */
732 25072 static float noise_enhancer(float fixed_gain, float *prev_tr_gain,
733 float voice_fac, float stab_fac)
734 {
735 25072 float sm_fac = 0.5 * (1 - voice_fac) * stab_fac;
736 float g0;
737
738 // XXX: the following fixed-point constants used to in(de)crement
739 // gain by 1.5dB were taken from the reference code, maybe it could
740 // be simpler
741
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 25072 if (fixed_gain < *prev_tr_gain) {
742
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 13115 g0 = FFMIN(*prev_tr_gain, fixed_gain + fixed_gain *
743 (6226 * (1.0f / (1 << 15)))); // +1.5 dB
744 } else
745
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 11957 g0 = FFMAX(*prev_tr_gain, fixed_gain *
746 (27536 * (1.0f / (1 << 15)))); // -1.5 dB
747
748 25072 *prev_tr_gain = g0; // update next frame threshold
749
750 25072 return sm_fac * g0 + (1 - sm_fac) * fixed_gain;
751 }
752
753 /**
754 * Filter the fixed_vector to emphasize the higher frequencies.
755 *
756 * @param[in,out] fixed_vector Fixed codebook vector
757 * @param[in] voice_fac Frame voicing factor
758 */
759 25072 static void pitch_enhancer(float *fixed_vector, float voice_fac)
760 {
761 int i;
762 25072 float cpe = 0.125 * (1 + voice_fac);
763 25072 float last = fixed_vector[0]; // holds c(i - 1)
764
765 25072 fixed_vector[0] -= cpe * fixed_vector[1];
766
767
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 1579536 for (i = 1; i < AMRWB_SFR_SIZE - 1; i++) {
768 1554464 float cur = fixed_vector[i];
769
770 1554464 fixed_vector[i] -= cpe * (last + fixed_vector[i + 1]);
771 1554464 last = cur;
772 }
773
774 25072 fixed_vector[AMRWB_SFR_SIZE - 1] -= cpe * last;
775 25072 }
776
777 /**
778 * Conduct 16th order linear predictive coding synthesis from excitation.
779 *
780 * @param[in] ctx Pointer to the AMRWBContext
781 * @param[in] lpc Pointer to the LPC coefficients
782 * @param[out] excitation Buffer for synthesis final excitation
783 * @param[in] fixed_gain Fixed codebook gain for synthesis
784 * @param[in] fixed_vector Algebraic codebook vector
785 * @param[in,out] samples Pointer to the output samples and memory
786 */
787 25072 static void synthesis(AMRWBContext *ctx, float *lpc, float *excitation,
788 float fixed_gain, const float *fixed_vector,
789 float *samples)
790 {
791 25072 ctx->acelpv_ctx.weighted_vector_sumf(excitation, ctx->pitch_vector, fixed_vector,
792 ctx->pitch_gain[0], fixed_gain, AMRWB_SFR_SIZE);
793
794 /* emphasize pitch vector contribution in low bitrate modes */
795
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 25072 if (ctx->pitch_gain[0] > 0.5 && ctx->fr_cur_mode <= MODE_8k85) {
796 int i;
797 1897 float energy = ctx->celpm_ctx.dot_productf(excitation, excitation,
798 AMRWB_SFR_SIZE);
799
800 // XXX: Weird part in both ref code and spec. A unknown parameter
801 // {beta} seems to be identical to the current pitch gain
802 1897 float pitch_factor = 0.25 * ctx->pitch_gain[0] * ctx->pitch_gain[0];
803
804
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 123305 for (i = 0; i < AMRWB_SFR_SIZE; i++)
805 121408 excitation[i] += pitch_factor * ctx->pitch_vector[i];
806
807 1897 ff_scale_vector_to_given_sum_of_squares(excitation, excitation,
808 energy, AMRWB_SFR_SIZE);
809 }
810
811 25072 ctx->celpf_ctx.celp_lp_synthesis_filterf(samples, lpc, excitation,
812 AMRWB_SFR_SIZE, LP_ORDER);
813 25072 }
814
815 /**
816 * Apply to synthesis a de-emphasis filter of the form:
817 * H(z) = 1 / (1 - m * z^-1)
818 *
819 * @param[out] out Output buffer
820 * @param[in] in Input samples array with in[-1]
821 * @param[in] m Filter coefficient
822 * @param[in,out] mem State from last filtering
823 */
824 25072 static void de_emphasis(float *out, float *in, float m, float mem[1])
825 {
826 int i;
827
828 25072 out[0] = in[0] + m * mem[0];
829
830
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 1604608 for (i = 1; i < AMRWB_SFR_SIZE; i++)
831 1579536 out[i] = in[i] + out[i - 1] * m;
832
833 25072 mem[0] = out[AMRWB_SFR_SIZE - 1];
834 25072 }
835
836 /**
837 * Upsample a signal by 5/4 ratio (from 12.8kHz to 16kHz) using
838 * a FIR interpolation filter. Uses past data from before *in address.
839 *
840 * @param[out] out Buffer for interpolated signal
841 * @param[in] in Current signal data (length 0.8*o_size)
842 * @param[in] o_size Output signal length
843 * @param[in] ctx The context
844 */
845 25072 static void upsample_5_4(float *out, const float *in, int o_size, CELPMContext *ctx)
846 {
847 25072 const float *in0 = in - UPS_FIR_SIZE + 1;
848 int i, j, k;
849 25072 int int_part = 0, frac_part;
850
851 25072 i = 0;
852
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 426224 for (j = 0; j < o_size / 5; j++) {
853 401152 out[i] = in[int_part];
854 401152 frac_part = 4;
855 401152 i++;
856
857
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 2005760 for (k = 1; k < 5; k++) {
858 3209216 out[i] = ctx->dot_productf(in0 + int_part,
859 1604608 upsample_fir[4 - frac_part],
860 UPS_MEM_SIZE);
861 1604608 int_part++;
862 1604608 frac_part--;
863 1604608 i++;
864 }
865 }
866 25072 }
867
868 /**
869 * Calculate the high-band gain based on encoded index (23k85 mode) or
870 * on the low-band speech signal and the Voice Activity Detection flag.
871 *
872 * @param[in] ctx The context
873 * @param[in] synth LB speech synthesis at 12.8k
874 * @param[in] hb_idx Gain index for mode 23k85 only
875 * @param[in] vad VAD flag for the frame
876 */
877 25072 static float find_hb_gain(AMRWBContext *ctx, const float *synth,
878 uint16_t hb_idx, uint8_t vad)
879 {
880 25072 int wsp = (vad > 0);
881 float tilt;
882 float tmp;
883
884
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 25072 if (ctx->fr_cur_mode == MODE_23k85)
885 8656 return qua_hb_gain[hb_idx] * (1.0f / (1 << 14));
886
887 16416 tmp = ctx->celpm_ctx.dot_productf(synth, synth + 1, AMRWB_SFR_SIZE - 1);
888
889
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 16416 if (tmp > 0) {
890 15653 tilt = tmp / ctx->celpm_ctx.dot_productf(synth, synth, AMRWB_SFR_SIZE);
891 } else
892 763 tilt = 0;
893
894 /* return gain bounded by [0.1, 1.0] */
895 16416 return av_clipf((1.0 - tilt) * (1.25 - 0.25 * wsp), 0.1, 1.0);
896 }
897
898 /**
899 * Generate the high-band excitation with the same energy from the lower
900 * one and scaled by the given gain.
901 *
902 * @param[in] ctx The context
903 * @param[out] hb_exc Buffer for the excitation
904 * @param[in] synth_exc Low-band excitation used for synthesis
905 * @param[in] hb_gain Wanted excitation gain
906 */
907 25072 static void scaled_hb_excitation(AMRWBContext *ctx, float *hb_exc,
908 const float *synth_exc, float hb_gain)
909 {
910 int i;
911 25072 float energy = ctx->celpm_ctx.dot_productf(synth_exc, synth_exc,
912 AMRWB_SFR_SIZE);
913
914 /* Generate a white-noise excitation */
915
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 2030832 for (i = 0; i < AMRWB_SFR_SIZE_16k; i++)
916 2005760 hb_exc[i] = 32768.0 - (uint16_t) av_lfg_get(&ctx->prng);
917
918 25072 ff_scale_vector_to_given_sum_of_squares(hb_exc, hb_exc,
919 25072 energy * hb_gain * hb_gain,
920 AMRWB_SFR_SIZE_16k);
921 25072 }
922
923 /**
924 * Calculate the auto-correlation for the ISF difference vector.
925 */
926 6156 static float auto_correlation(float *diff_isf, float mean, int lag)
927 {
928 int i;
929 6156 float sum = 0.0;
930
931
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 49248 for (i = 7; i < LP_ORDER - 2; i++) {
932 43092 float prod = (diff_isf[i] - mean) * (diff_isf[i - lag] - mean);
933 43092 sum += prod * prod;
934 }
935 6156 return sum;
936 }
937
938 /**
939 * Extrapolate a ISF vector to the 16kHz range (20th order LP)
940 * used at mode 6k60 LP filter for the high frequency band.
941 *
942 * @param[out] isf Buffer for extrapolated isf; contains LP_ORDER
943 * values on input
944 */
945 2052 static void extrapolate_isf(float isf[LP_ORDER_16k])
946 {
947 float diff_isf[LP_ORDER - 2], diff_mean;
948 float corr_lag[3];
949 float est, scale;
950 int i, j, i_max_corr;
951
952 2052 isf[LP_ORDER_16k - 1] = isf[LP_ORDER - 1];
953
954 /* Calculate the difference vector */
955
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 30780 for (i = 0; i < LP_ORDER - 2; i++)
956 28728 diff_isf[i] = isf[i + 1] - isf[i];
957
958 2052 diff_mean = 0.0;
959
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 26676 for (i = 2; i < LP_ORDER - 2; i++)
960 24624 diff_mean += diff_isf[i] * (1.0f / (LP_ORDER - 4));
961
962 /* Find which is the maximum autocorrelation */
963 2052 i_max_corr = 0;
964
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 8208 for (i = 0; i < 3; i++) {
965 6156 corr_lag[i] = auto_correlation(diff_isf, diff_mean, i + 2);
966
967
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 6156 if (corr_lag[i] > corr_lag[i_max_corr])
968 2154 i_max_corr = i;
969 }
970 2052 i_max_corr++;
971
972
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 10260 for (i = LP_ORDER - 1; i < LP_ORDER_16k - 1; i++)
973 8208 isf[i] = isf[i - 1] + isf[i - 1 - i_max_corr]
974 8208 - isf[i - 2 - i_max_corr];
975
976 /* Calculate an estimate for ISF(18) and scale ISF based on the error */
977 2052 est = 7965 + (isf[2] - isf[3] - isf[4]) / 6.0;
978
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 2052 scale = 0.5 * (FFMIN(est, 7600) - isf[LP_ORDER - 2]) /
979 2052 (isf[LP_ORDER_16k - 2] - isf[LP_ORDER - 2]);
980
981
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 10260 for (i = LP_ORDER - 1, j = 0; i < LP_ORDER_16k - 1; i++, j++)
982 8208 diff_isf[j] = scale * (isf[i] - isf[i - 1]);
983
984 /* Stability insurance */
985
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 8208 for (i = 1; i < LP_ORDER_16k - LP_ORDER; i++)
986
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 6156 if (diff_isf[i] + diff_isf[i - 1] < 5.0) {
987 ✗ if (diff_isf[i] > diff_isf[i - 1]) {
988 ✗ diff_isf[i - 1] = 5.0 - diff_isf[i];
989 } else
990 ✗ diff_isf[i] = 5.0 - diff_isf[i - 1];
991 }
992
993
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 10260 for (i = LP_ORDER - 1, j = 0; i < LP_ORDER_16k - 1; i++, j++)
994 8208 isf[i] = isf[i - 1] + diff_isf[j] * (1.0f / (1 << 15));
995
996 /* Scale the ISF vector for 16000 Hz */
997
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 41040 for (i = 0; i < LP_ORDER_16k - 1; i++)
998 38988 isf[i] *= 0.8;
999 2052 }
1000
1001 /**
1002 * Spectral expand the LP coefficients using the equation:
1003 * y[i] = x[i] * (gamma ** i)
1004 *
1005 * @param[out] out Output buffer (may use input array)
1006 * @param[in] lpc LP coefficients array
1007 * @param[in] gamma Weighting factor
1008 * @param[in] size LP array size
1009 */
1010 25072 static void lpc_weighting(float *out, const float *lpc, float gamma, int size)
1011 {
1012 int i;
1013 25072 float fac = gamma;
1014
1015
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 434432 for (i = 0; i < size; i++) {
1016 409360 out[i] = lpc[i] * fac;
1017 409360 fac *= gamma;
1018 }
1019 25072 }
1020
1021 /**
1022 * Conduct 20th order linear predictive coding synthesis for the high
1023 * frequency band excitation at 16kHz.
1024 *
1025 * @param[in] ctx The context
1026 * @param[in] subframe Current subframe index (0 to 3)
1027 * @param[in,out] samples Pointer to the output speech samples
1028 * @param[in] exc Generated white-noise scaled excitation
1029 * @param[in] isf Current frame isf vector
1030 * @param[in] isf_past Past frame final isf vector
1031 */
1032 25072 static void hb_synthesis(AMRWBContext *ctx, int subframe, float *samples,
1033 const float *exc, const float *isf, const float *isf_past)
1034 {
1035 float hb_lpc[LP_ORDER_16k];
1036 25072 enum Mode mode = ctx->fr_cur_mode;
1037
1038
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 25072 if (mode == MODE_6k60) {
1039 float e_isf[LP_ORDER_16k]; // ISF vector for extrapolation
1040 double e_isp[LP_ORDER_16k];
1041
1042 2052 ctx->acelpv_ctx.weighted_vector_sumf(e_isf, isf_past, isf, isfp_inter[subframe],
1043 2052 1.0 - isfp_inter[subframe], LP_ORDER);
1044
1045 2052 extrapolate_isf(e_isf);
1046
1047 2052 e_isf[LP_ORDER_16k - 1] *= 2.0;
1048 2052 ff_acelp_lsf2lspd(e_isp, e_isf, LP_ORDER_16k);
1049 2052 ff_amrwb_lsp2lpc(e_isp, hb_lpc, LP_ORDER_16k);
1050
1051 2052 lpc_weighting(hb_lpc, hb_lpc, 0.9, LP_ORDER_16k);
1052 } else {
1053 23020 lpc_weighting(hb_lpc, ctx->lp_coef[subframe], 0.6, LP_ORDER);
1054 }
1055
1056
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 25072 ctx->celpf_ctx.celp_lp_synthesis_filterf(samples, hb_lpc, exc, AMRWB_SFR_SIZE_16k,
1057 (mode == MODE_6k60) ? LP_ORDER_16k : LP_ORDER);
1058 25072 }
1059
1060 /**
1061 * Apply a 15th order filter to high-band samples.
1062 * The filter characteristic depends on the given coefficients.
1063 *
1064 * @param[out] out Buffer for filtered output
1065 * @param[in] fir_coef Filter coefficients
1066 * @param[in,out] mem State from last filtering (updated)
1067 * @param[in] in Input speech data (high-band)
1068 *
1069 * @remark It is safe to pass the same array in in and out parameters
1070 */
1071
1072 #ifndef hb_fir_filter
1073 33728 static void hb_fir_filter(float *out, const float fir_coef[HB_FIR_SIZE + 1],
1074 float mem[HB_FIR_SIZE], const float *in)
1075 {
1076 int i, j;
1077 float data[AMRWB_SFR_SIZE_16k + HB_FIR_SIZE]; // past and current samples
1078
1079 33728 memcpy(data, mem, HB_FIR_SIZE * sizeof(float));
1080 33728 memcpy(data + HB_FIR_SIZE, in, AMRWB_SFR_SIZE_16k * sizeof(float));
1081
1082
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 2731968 for (i = 0; i < AMRWB_SFR_SIZE_16k; i++) {
1083 2698240 out[i] = 0.0;
1084
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 86343680 for (j = 0; j <= HB_FIR_SIZE; j++)
1085 83645440 out[i] += data[i + j] * fir_coef[j];
1086 }
1087
1088 33728 memcpy(mem, data + AMRWB_SFR_SIZE_16k, HB_FIR_SIZE * sizeof(float));
1089 33728 }
1090 #endif /* hb_fir_filter */
1091
1092 /**
1093 * Update context state before the next subframe.
1094 */
1095 25072 static void update_sub_state(AMRWBContext *ctx)
1096 {
1097 25072 memmove(&ctx->excitation_buf[0], &ctx->excitation_buf[AMRWB_SFR_SIZE],
1098 (AMRWB_P_DELAY_MAX + LP_ORDER + 1) * sizeof(float));
1099
1100 25072 memmove(&ctx->pitch_gain[1], &ctx->pitch_gain[0], 5 * sizeof(float));
1101 25072 memmove(&ctx->fixed_gain[1], &ctx->fixed_gain[0], 1 * sizeof(float));
1102
1103 25072 memmove(&ctx->samples_az[0], &ctx->samples_az[AMRWB_SFR_SIZE],
1104 LP_ORDER * sizeof(float));
1105 25072 memmove(&ctx->samples_up[0], &ctx->samples_up[AMRWB_SFR_SIZE],
1106 UPS_MEM_SIZE * sizeof(float));
1107 25072 memmove(&ctx->samples_hb[0], &ctx->samples_hb[AMRWB_SFR_SIZE_16k],
1108 LP_ORDER_16k * sizeof(float));
1109 25072 }
1110
1111 6268 static int amrwb_decode_frame(AVCodecContext *avctx, AVFrame *frame,
1112 int *got_frame_ptr, AVPacket *avpkt)
1113 {
1114 6268 AMRWBChannelsContext *s = avctx->priv_data;
1115 6268 const uint8_t *buf = avpkt->data;
1116 6268 int buf_size = avpkt->size;
1117 int sub, i, ret;
1118
1119 /* get output buffer */
1120 6268 frame->nb_samples = 4 * AMRWB_SFR_SIZE_16k;
1121
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 6268 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
1122 ✗ return ret;
1123
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 12536 for (int ch = 0; ch < avctx->ch_layout.nb_channels; ch++) {
1125 6268 AMRWBContext *ctx = &s->ch[ch];
1126 6268 AMRWBFrame *cf = &ctx->frame;
1127 int expected_fr_size, header_size;
1128 float spare_vector[AMRWB_SFR_SIZE]; // extra stack space to hold result from anti-sparseness processing
1129 float fixed_gain_factor; // fixed gain correction factor (gamma)
1130 float *synth_fixed_vector; // pointer to the fixed vector that synthesis should use
1131 float synth_fixed_gain; // the fixed gain that synthesis should use
1132 float voice_fac, stab_fac; // parameters used for gain smoothing
1133 float synth_exc[AMRWB_SFR_SIZE]; // post-processed excitation for synthesis
1134 float hb_exc[AMRWB_SFR_SIZE_16k]; // excitation for the high frequency band
1135 float hb_samples[AMRWB_SFR_SIZE_16k]; // filtered high-band samples from synthesis
1136 float hb_gain;
1137 6268 float *buf_out = (float *)frame->extended_data[ch];
1138
1139 6268 header_size = decode_mime_header(ctx, buf);
1140 6268 expected_fr_size = ((cf_sizes_wb[ctx->fr_cur_mode] + 7) >> 3) + 1;
1141
1142
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 6268 if (!ctx->fr_quality)
1143 ✗ av_log(avctx, AV_LOG_ERROR, "Encountered a bad or corrupted frame\n");
1144
1145
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 6268 if (ctx->fr_cur_mode == NO_DATA || !ctx->fr_quality) {
1146 /* The specification suggests a "random signal" and
1147 "a muting technique" to "gradually decrease the output level". */
1148 ✗ av_samples_set_silence(&frame->extended_data[ch], 0, frame->nb_samples, 1, AV_SAMPLE_FMT_FLT);
1149 ✗ buf += expected_fr_size;
1150 ✗ buf_size -= expected_fr_size;
1151 ✗ continue;
1152 }
1153
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 6268 if (ctx->fr_cur_mode > MODE_SID) {
1154 ✗ av_log(avctx, AV_LOG_ERROR,
1155 ✗ "Invalid mode %d\n", ctx->fr_cur_mode);
1156 ✗ return AVERROR_INVALIDDATA;
1157 }
1158
1159
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 6268 if (buf_size < expected_fr_size) {
1160 ✗ av_log(avctx, AV_LOG_ERROR,
1161 "Frame too small (%d bytes). Truncated file?\n", buf_size);
1162 ✗ *got_frame_ptr = 0;
1163 ✗ return AVERROR_INVALIDDATA;
1164 }
1165
1166
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 6268 if (ctx->fr_cur_mode == MODE_SID) { /* Comfort noise frame */
1167 ✗ avpriv_request_sample(avctx, "SID mode");
1168 ✗ return AVERROR_PATCHWELCOME;
1169 }
1170
1171 6268 ff_amr_bit_reorder((uint16_t *) &ctx->frame, sizeof(AMRWBFrame),
1172 6268 buf + header_size, amr_bit_orderings_by_mode[ctx->fr_cur_mode]);
1173
1174 /* Decode the quantized ISF vector */
1175
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 6268 if (ctx->fr_cur_mode == MODE_6k60) {
1176 513 decode_isf_indices_36b(cf->isp_id, ctx->isf_cur);
1177 } else {
1178 5755 decode_isf_indices_46b(cf->isp_id, ctx->isf_cur);
1179 }
1180
1181 6268 isf_add_mean_and_past(ctx->isf_cur, ctx->isf_q_past);
1182 6268 ff_set_min_dist_lsf(ctx->isf_cur, MIN_ISF_SPACING, LP_ORDER - 1);
1183
1184 6268 stab_fac = stability_factor(ctx->isf_cur, ctx->isf_past_final);
1185
1186 6268 ctx->isf_cur[LP_ORDER - 1] *= 2.0;
1187 6268 ff_acelp_lsf2lspd(ctx->isp[3], ctx->isf_cur, LP_ORDER);
1188
1189 /* Generate a ISP vector for each subframe */
1190
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 6268 if (ctx->first_frame) {
1191 22 ctx->first_frame = 0;
1192 22 memcpy(ctx->isp_sub4_past, ctx->isp[3], LP_ORDER * sizeof(double));
1193 }
1194 6268 interpolate_isp(ctx->isp, ctx->isp_sub4_past);
1195
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 31340 for (sub = 0; sub < 4; sub++)
1197 25072 ff_amrwb_lsp2lpc(ctx->isp[sub], ctx->lp_coef[sub], LP_ORDER);
1198
1199
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 31340 for (sub = 0; sub < 4; sub++) {
1200 25072 const AMRWBSubFrame *cur_subframe = &cf->subframe[sub];
1201 25072 float *sub_buf = buf_out + sub * AMRWB_SFR_SIZE_16k;
1202
1203 /* Decode adaptive codebook (pitch vector) */
1204 25072 decode_pitch_vector(ctx, cur_subframe, sub);
1205 /* Decode innovative codebook (fixed vector) */
1206 25072 decode_fixed_vector(ctx->fixed_vector, cur_subframe->pul_ih,
1207 25072 cur_subframe->pul_il, ctx->fr_cur_mode);
1208
1209 25072 pitch_sharpening(ctx, ctx->fixed_vector);
1210
1211 25072 decode_gains(cur_subframe->vq_gain, ctx->fr_cur_mode,
1212 &fixed_gain_factor, &ctx->pitch_gain[0]);
1213
1214 25072 ctx->fixed_gain[0] =
1215 25072 ff_amr_set_fixed_gain(fixed_gain_factor,
1216 25072 ctx->celpm_ctx.dot_productf(ctx->fixed_vector,
1217 25072 ctx->fixed_vector,
1218 AMRWB_SFR_SIZE) /
1219 AMRWB_SFR_SIZE,
1220 25072 ctx->prediction_error,
1221 ENERGY_MEAN, energy_pred_fac);
1222
1223 /* Calculate voice factor and store tilt for next subframe */
1224 25072 voice_fac = voice_factor(ctx->pitch_vector, ctx->pitch_gain[0],
1225 25072 ctx->fixed_vector, ctx->fixed_gain[0],
1226 &ctx->celpm_ctx);
1227 25072 ctx->tilt_coef = voice_fac * 0.25 + 0.25;
1228
1229 /* Construct current excitation */
1230
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 1629680 for (i = 0; i < AMRWB_SFR_SIZE; i++) {
1231 1604608 ctx->excitation[i] *= ctx->pitch_gain[0];
1232 1604608 ctx->excitation[i] += ctx->fixed_gain[0] * ctx->fixed_vector[i];
1233 1604608 ctx->excitation[i] = truncf(ctx->excitation[i]);
1234 }
1235
1236 /* Post-processing of excitation elements */
1237 25072 synth_fixed_gain = noise_enhancer(ctx->fixed_gain[0], &ctx->prev_tr_gain,
1238 voice_fac, stab_fac);
1239
1240 25072 synth_fixed_vector = anti_sparseness(ctx, ctx->fixed_vector,
1241 spare_vector);
1242
1243 25072 pitch_enhancer(synth_fixed_vector, voice_fac);
1244
1245 25072 synthesis(ctx, ctx->lp_coef[sub], synth_exc, synth_fixed_gain,
1246 synth_fixed_vector, &ctx->samples_az[LP_ORDER]);
1247
1248 /* Synthesis speech post-processing */
1249 25072 de_emphasis(&ctx->samples_up[UPS_MEM_SIZE],
1250 25072 &ctx->samples_az[LP_ORDER], PREEMPH_FAC, ctx->demph_mem);
1251
1252 25072 ctx->acelpf_ctx.acelp_apply_order_2_transfer_function(&ctx->samples_up[UPS_MEM_SIZE],
1253 25072 &ctx->samples_up[UPS_MEM_SIZE], hpf_zeros, hpf_31_poles,
1254 25072 hpf_31_gain, ctx->hpf_31_mem, AMRWB_SFR_SIZE);
1255
1256 25072 upsample_5_4(sub_buf, &ctx->samples_up[UPS_FIR_SIZE],
1257 AMRWB_SFR_SIZE_16k, &ctx->celpm_ctx);
1258
1259 /* High frequency band (6.4 - 7.0 kHz) generation part */
1260 25072 ctx->acelpf_ctx.acelp_apply_order_2_transfer_function(hb_samples,
1261 25072 &ctx->samples_up[UPS_MEM_SIZE], hpf_zeros, hpf_400_poles,
1262 25072 hpf_400_gain, ctx->hpf_400_mem, AMRWB_SFR_SIZE);
1263
1264 25072 hb_gain = find_hb_gain(ctx, hb_samples,
1265 25072 cur_subframe->hb_gain, cf->vad);
1266
1267 25072 scaled_hb_excitation(ctx, hb_exc, synth_exc, hb_gain);
1268
1269 25072 hb_synthesis(ctx, sub, &ctx->samples_hb[LP_ORDER_16k],
1270 25072 hb_exc, ctx->isf_cur, ctx->isf_past_final);
1271
1272 /* High-band post-processing filters */
1273 25072 hb_fir_filter(hb_samples, bpf_6_7_coef, ctx->bpf_6_7_mem,
1274 25072 &ctx->samples_hb[LP_ORDER_16k]);
1275
1276
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 25072 if (ctx->fr_cur_mode == MODE_23k85)
1277 8656 hb_fir_filter(hb_samples, lpf_7_coef, ctx->lpf_7_mem,
1278 hb_samples);
1279
1280 /* Add the low and high frequency bands */
1281
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 2030832 for (i = 0; i < AMRWB_SFR_SIZE_16k; i++)
1282 2005760 sub_buf[i] = (sub_buf[i] + hb_samples[i]) * (1.0f / (1 << 15));
1283
1284 /* Update buffers and history */
1285 25072 update_sub_state(ctx);
1286 }
1287
1288 /* update state for next frame */
1289 6268 memcpy(ctx->isp_sub4_past, ctx->isp[3], LP_ORDER * sizeof(ctx->isp[3][0]));
1290 6268 memcpy(ctx->isf_past_final, ctx->isf_cur, LP_ORDER * sizeof(float));
1291
1292 6268 buf += expected_fr_size;
1293 6268 buf_size -= expected_fr_size;
1294 }
1295
1296 6268 *got_frame_ptr = 1;
1297
1298 6268 return buf - avpkt->data;
1299 }
1300
1301 const FFCodec ff_amrwb_decoder = {
1302 .p.name = "amrwb",
1303 CODEC_LONG_NAME("AMR-WB (Adaptive Multi-Rate WideBand)"),
1304 .p.type = AVMEDIA_TYPE_AUDIO,
1305 .p.id = AV_CODEC_ID_AMR_WB,
1306 .priv_data_size = sizeof(AMRWBChannelsContext),
1307 .init = amrwb_decode_init,
1308 FF_CODEC_DECODE_CB(amrwb_decode_frame),
1309 .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF,
1310 .p.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP,
1311 AV_SAMPLE_FMT_NONE },
1312 };
1313