Imported Debian version 2.5.3~trusty1
[deb_ffmpeg.git] / ffmpeg / libavcodec / amrwbdec.c
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DM
1/*
2 * AMR wideband decoder
3 * Copyright (c) 2010 Marcelo Galvao Povoa
4 *
5 * This file is part of FFmpeg.
6 *
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
11 *
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A particular PURPOSE. See the GNU
15 * Lesser General Public License for more details.
16 *
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20 */
21
22/**
23 * @file
24 * AMR wideband decoder
25 */
26
27#include "libavutil/channel_layout.h"
28#include "libavutil/common.h"
29#include "libavutil/float_dsp.h"
30#include "libavutil/lfg.h"
31
32#include "avcodec.h"
33#include "lsp.h"
34#include "celp_filters.h"
35#include "celp_math.h"
36#include "acelp_filters.h"
37#include "acelp_vectors.h"
38#include "acelp_pitch_delay.h"
39#include "internal.h"
40
41#define AMR_USE_16BIT_TABLES
42#include "amr.h"
43
44#include "amrwbdata.h"
45#include "mips/amrwbdec_mips.h"
46
47typedef struct {
48 AMRWBFrame frame; ///< AMRWB parameters decoded from bitstream
49 enum Mode fr_cur_mode; ///< mode index of current frame
50 uint8_t fr_quality; ///< frame quality index (FQI)
51 float isf_cur[LP_ORDER]; ///< working ISF vector from current frame
52 float isf_q_past[LP_ORDER]; ///< quantized ISF vector of the previous frame
53 float isf_past_final[LP_ORDER]; ///< final processed ISF vector of the previous frame
54 double isp[4][LP_ORDER]; ///< ISP vectors from current frame
55 double isp_sub4_past[LP_ORDER]; ///< ISP vector for the 4th subframe of the previous frame
56
57 float lp_coef[4][LP_ORDER]; ///< Linear Prediction Coefficients from ISP vector
58
59 uint8_t base_pitch_lag; ///< integer part of pitch lag for the next relative subframe
60 uint8_t pitch_lag_int; ///< integer part of pitch lag of the previous subframe
61
62 float excitation_buf[AMRWB_P_DELAY_MAX + LP_ORDER + 2 + AMRWB_SFR_SIZE]; ///< current excitation and all necessary excitation history
63 float *excitation; ///< points to current excitation in excitation_buf[]
64
65 float pitch_vector[AMRWB_SFR_SIZE]; ///< adaptive codebook (pitch) vector for current subframe
66 float fixed_vector[AMRWB_SFR_SIZE]; ///< algebraic codebook (fixed) vector for current subframe
67
68 float prediction_error[4]; ///< quantified prediction errors {20log10(^gamma_gc)} for previous four subframes
69 float pitch_gain[6]; ///< quantified pitch gains for the current and previous five subframes
70 float fixed_gain[2]; ///< quantified fixed gains for the current and previous subframes
71
72 float tilt_coef; ///< {beta_1} related to the voicing of the previous subframe
73
74 float prev_sparse_fixed_gain; ///< previous fixed gain; used by anti-sparseness to determine "onset"
75 uint8_t prev_ir_filter_nr; ///< previous impulse response filter "impNr": 0 - strong, 1 - medium, 2 - none
76 float prev_tr_gain; ///< previous initial gain used by noise enhancer for threshold
77
78 float samples_az[LP_ORDER + AMRWB_SFR_SIZE]; ///< low-band samples and memory from synthesis at 12.8kHz
79 float samples_up[UPS_MEM_SIZE + AMRWB_SFR_SIZE]; ///< low-band samples and memory processed for upsampling
80 float samples_hb[LP_ORDER_16k + AMRWB_SFR_SIZE_16k]; ///< high-band samples and memory from synthesis at 16kHz
81
82 float hpf_31_mem[2], hpf_400_mem[2]; ///< previous values in the high pass filters
83 float demph_mem[1]; ///< previous value in the de-emphasis filter
84 float bpf_6_7_mem[HB_FIR_SIZE]; ///< previous values in the high-band band pass filter
85 float lpf_7_mem[HB_FIR_SIZE]; ///< previous values in the high-band low pass filter
86
87 AVLFG prng; ///< random number generator for white noise excitation
88 uint8_t first_frame; ///< flag active during decoding of the first frame
89 ACELPFContext acelpf_ctx; ///< context for filters for ACELP-based codecs
90 ACELPVContext acelpv_ctx; ///< context for vector operations for ACELP-based codecs
91 CELPFContext celpf_ctx; ///< context for filters for CELP-based codecs
92 CELPMContext celpm_ctx; ///< context for fixed point math operations
93
94} AMRWBContext;
95
96static av_cold int amrwb_decode_init(AVCodecContext *avctx)
97{
98 AMRWBContext *ctx = avctx->priv_data;
99 int i;
100
101 if (avctx->channels > 1) {
102 avpriv_report_missing_feature(avctx, "multi-channel AMR");
103 return AVERROR_PATCHWELCOME;
104 }
105
106 avctx->channels = 1;
107 avctx->channel_layout = AV_CH_LAYOUT_MONO;
108 if (!avctx->sample_rate)
109 avctx->sample_rate = 16000;
110 avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
111
112 av_lfg_init(&ctx->prng, 1);
113
114 ctx->excitation = &ctx->excitation_buf[AMRWB_P_DELAY_MAX + LP_ORDER + 1];
115 ctx->first_frame = 1;
116
117 for (i = 0; i < LP_ORDER; i++)
118 ctx->isf_past_final[i] = isf_init[i] * (1.0f / (1 << 15));
119
120 for (i = 0; i < 4; i++)
121 ctx->prediction_error[i] = MIN_ENERGY;
122
123 ff_acelp_filter_init(&ctx->acelpf_ctx);
124 ff_acelp_vectors_init(&ctx->acelpv_ctx);
125 ff_celp_filter_init(&ctx->celpf_ctx);
126 ff_celp_math_init(&ctx->celpm_ctx);
127
128 return 0;
129}
130
131/**
132 * Decode the frame header in the "MIME/storage" format. This format
133 * is simpler and does not carry the auxiliary frame information.
134 *
135 * @param[in] ctx The Context
136 * @param[in] buf Pointer to the input buffer
137 *
138 * @return The decoded header length in bytes
139 */
140static int decode_mime_header(AMRWBContext *ctx, const uint8_t *buf)
141{
142 /* Decode frame header (1st octet) */
143 ctx->fr_cur_mode = buf[0] >> 3 & 0x0F;
144 ctx->fr_quality = (buf[0] & 0x4) == 0x4;
145
146 return 1;
147}
148
149/**
150 * Decode quantized ISF vectors using 36-bit indexes (6K60 mode only).
151 *
152 * @param[in] ind Array of 5 indexes
153 * @param[out] isf_q Buffer for isf_q[LP_ORDER]
154 *
155 */
156static void decode_isf_indices_36b(uint16_t *ind, float *isf_q)
157{
158 int i;
159
160 for (i = 0; i < 9; i++)
161 isf_q[i] = dico1_isf[ind[0]][i] * (1.0f / (1 << 15));
162
163 for (i = 0; i < 7; i++)
164 isf_q[i + 9] = dico2_isf[ind[1]][i] * (1.0f / (1 << 15));
165
166 for (i = 0; i < 5; i++)
167 isf_q[i] += dico21_isf_36b[ind[2]][i] * (1.0f / (1 << 15));
168
169 for (i = 0; i < 4; i++)
170 isf_q[i + 5] += dico22_isf_36b[ind[3]][i] * (1.0f / (1 << 15));
171
172 for (i = 0; i < 7; i++)
173 isf_q[i + 9] += dico23_isf_36b[ind[4]][i] * (1.0f / (1 << 15));
174}
175
176/**
177 * Decode quantized ISF vectors using 46-bit indexes (except 6K60 mode).
178 *
179 * @param[in] ind Array of 7 indexes
180 * @param[out] isf_q Buffer for isf_q[LP_ORDER]
181 *
182 */
183static void decode_isf_indices_46b(uint16_t *ind, float *isf_q)
184{
185 int i;
186
187 for (i = 0; i < 9; i++)
188 isf_q[i] = dico1_isf[ind[0]][i] * (1.0f / (1 << 15));
189
190 for (i = 0; i < 7; i++)
191 isf_q[i + 9] = dico2_isf[ind[1]][i] * (1.0f / (1 << 15));
192
193 for (i = 0; i < 3; i++)
194 isf_q[i] += dico21_isf[ind[2]][i] * (1.0f / (1 << 15));
195
196 for (i = 0; i < 3; i++)
197 isf_q[i + 3] += dico22_isf[ind[3]][i] * (1.0f / (1 << 15));
198
199 for (i = 0; i < 3; i++)
200 isf_q[i + 6] += dico23_isf[ind[4]][i] * (1.0f / (1 << 15));
201
202 for (i = 0; i < 3; i++)
203 isf_q[i + 9] += dico24_isf[ind[5]][i] * (1.0f / (1 << 15));
204
205 for (i = 0; i < 4; i++)
206 isf_q[i + 12] += dico25_isf[ind[6]][i] * (1.0f / (1 << 15));
207}
208
209/**
210 * Apply mean and past ISF values using the prediction factor.
211 * Updates past ISF vector.
212 *
213 * @param[in,out] isf_q Current quantized ISF
214 * @param[in,out] isf_past Past quantized ISF
215 *
216 */
217static void isf_add_mean_and_past(float *isf_q, float *isf_past)
218{
219 int i;
220 float tmp;
221
222 for (i = 0; i < LP_ORDER; i++) {
223 tmp = isf_q[i];
224 isf_q[i] += isf_mean[i] * (1.0f / (1 << 15));
225 isf_q[i] += PRED_FACTOR * isf_past[i];
226 isf_past[i] = tmp;
227 }
228}
229
230/**
231 * Interpolate the fourth ISP vector from current and past frames
232 * to obtain an ISP vector for each subframe.
233 *
234 * @param[in,out] isp_q ISPs for each subframe
235 * @param[in] isp4_past Past ISP for subframe 4
236 */
237static void interpolate_isp(double isp_q[4][LP_ORDER], const double *isp4_past)
238{
239 int i, k;
240
241 for (k = 0; k < 3; k++) {
242 float c = isfp_inter[k];
243 for (i = 0; i < LP_ORDER; i++)
244 isp_q[k][i] = (1.0 - c) * isp4_past[i] + c * isp_q[3][i];
245 }
246}
247
248/**
249 * Decode an adaptive codebook index into pitch lag (except 6k60, 8k85 modes).
250 * Calculate integer lag and fractional lag always using 1/4 resolution.
251 * In 1st and 3rd subframes the index is relative to last subframe integer lag.
252 *
253 * @param[out] lag_int Decoded integer pitch lag
254 * @param[out] lag_frac Decoded fractional pitch lag
255 * @param[in] pitch_index Adaptive codebook pitch index
256 * @param[in,out] base_lag_int Base integer lag used in relative subframes
257 * @param[in] subframe Current subframe index (0 to 3)
258 */
259static void decode_pitch_lag_high(int *lag_int, int *lag_frac, int pitch_index,
260 uint8_t *base_lag_int, int subframe)
261{
262 if (subframe == 0 || subframe == 2) {
263 if (pitch_index < 376) {
264 *lag_int = (pitch_index + 137) >> 2;
265 *lag_frac = pitch_index - (*lag_int << 2) + 136;
266 } else if (pitch_index < 440) {
267 *lag_int = (pitch_index + 257 - 376) >> 1;
268 *lag_frac = (pitch_index - (*lag_int << 1) + 256 - 376) << 1;
269 /* the actual resolution is 1/2 but expressed as 1/4 */
270 } else {
271 *lag_int = pitch_index - 280;
272 *lag_frac = 0;
273 }
274 /* minimum lag for next subframe */
275 *base_lag_int = av_clip(*lag_int - 8 - (*lag_frac < 0),
276 AMRWB_P_DELAY_MIN, AMRWB_P_DELAY_MAX - 15);
277 // XXX: the spec states clearly that *base_lag_int should be
278 // the nearest integer to *lag_int (minus 8), but the ref code
279 // actually always uses its floor, I'm following the latter
280 } else {
281 *lag_int = (pitch_index + 1) >> 2;
282 *lag_frac = pitch_index - (*lag_int << 2);
283 *lag_int += *base_lag_int;
284 }
285}
286
287/**
288 * Decode an adaptive codebook index into pitch lag for 8k85 and 6k60 modes.
289 * The description is analogous to decode_pitch_lag_high, but in 6k60 the
290 * relative index is used for all subframes except the first.
291 */
292static void decode_pitch_lag_low(int *lag_int, int *lag_frac, int pitch_index,
293 uint8_t *base_lag_int, int subframe, enum Mode mode)
294{
295 if (subframe == 0 || (subframe == 2 && mode != MODE_6k60)) {
296 if (pitch_index < 116) {
297 *lag_int = (pitch_index + 69) >> 1;
298 *lag_frac = (pitch_index - (*lag_int << 1) + 68) << 1;
299 } else {
300 *lag_int = pitch_index - 24;
301 *lag_frac = 0;
302 }
303 // XXX: same problem as before
304 *base_lag_int = av_clip(*lag_int - 8 - (*lag_frac < 0),
305 AMRWB_P_DELAY_MIN, AMRWB_P_DELAY_MAX - 15);
306 } else {
307 *lag_int = (pitch_index + 1) >> 1;
308 *lag_frac = (pitch_index - (*lag_int << 1)) << 1;
309 *lag_int += *base_lag_int;
310 }
311}
312
313/**
314 * Find the pitch vector by interpolating the past excitation at the
315 * pitch delay, which is obtained in this function.
316 *
317 * @param[in,out] ctx The context
318 * @param[in] amr_subframe Current subframe data
319 * @param[in] subframe Current subframe index (0 to 3)
320 */
321static void decode_pitch_vector(AMRWBContext *ctx,
322 const AMRWBSubFrame *amr_subframe,
323 const int subframe)
324{
325 int pitch_lag_int, pitch_lag_frac;
326 int i;
327 float *exc = ctx->excitation;
328 enum Mode mode = ctx->fr_cur_mode;
329
330 if (mode <= MODE_8k85) {
331 decode_pitch_lag_low(&pitch_lag_int, &pitch_lag_frac, amr_subframe->adap,
332 &ctx->base_pitch_lag, subframe, mode);
333 } else
334 decode_pitch_lag_high(&pitch_lag_int, &pitch_lag_frac, amr_subframe->adap,
335 &ctx->base_pitch_lag, subframe);
336
337 ctx->pitch_lag_int = pitch_lag_int;
338 pitch_lag_int += pitch_lag_frac > 0;
339
340 /* Calculate the pitch vector by interpolating the past excitation at the
341 pitch lag using a hamming windowed sinc function */
342 ctx->acelpf_ctx.acelp_interpolatef(exc,
343 exc + 1 - pitch_lag_int,
344 ac_inter, 4,
345 pitch_lag_frac + (pitch_lag_frac > 0 ? 0 : 4),
346 LP_ORDER, AMRWB_SFR_SIZE + 1);
347
348 /* Check which pitch signal path should be used
349 * 6k60 and 8k85 modes have the ltp flag set to 0 */
350 if (amr_subframe->ltp) {
351 memcpy(ctx->pitch_vector, exc, AMRWB_SFR_SIZE * sizeof(float));
352 } else {
353 for (i = 0; i < AMRWB_SFR_SIZE; i++)
354 ctx->pitch_vector[i] = 0.18 * exc[i - 1] + 0.64 * exc[i] +
355 0.18 * exc[i + 1];
356 memcpy(exc, ctx->pitch_vector, AMRWB_SFR_SIZE * sizeof(float));
357 }
358}
359
360/** Get x bits in the index interval [lsb,lsb+len-1] inclusive */
361#define BIT_STR(x,lsb,len) (((x) >> (lsb)) & ((1 << (len)) - 1))
362
363/** Get the bit at specified position */
364#define BIT_POS(x, p) (((x) >> (p)) & 1)
365
366/**
367 * The next six functions decode_[i]p_track decode exactly i pulses
368 * positions and amplitudes (-1 or 1) in a subframe track using
369 * an encoded pulse indexing (TS 26.190 section 5.8.2).
370 *
371 * The results are given in out[], in which a negative number means
372 * amplitude -1 and vice versa (i.e., ampl(x) = x / abs(x) ).
373 *
374 * @param[out] out Output buffer (writes i elements)
375 * @param[in] code Pulse index (no. of bits varies, see below)
376 * @param[in] m (log2) Number of potential positions
377 * @param[in] off Offset for decoded positions
378 */
379static inline void decode_1p_track(int *out, int code, int m, int off)
380{
381 int pos = BIT_STR(code, 0, m) + off; ///code: m+1 bits
382
383 out[0] = BIT_POS(code, m) ? -pos : pos;
384}
385
386static inline void decode_2p_track(int *out, int code, int m, int off) ///code: 2m+1 bits
387{
388 int pos0 = BIT_STR(code, m, m) + off;
389 int pos1 = BIT_STR(code, 0, m) + off;
390
391 out[0] = BIT_POS(code, 2*m) ? -pos0 : pos0;
392 out[1] = BIT_POS(code, 2*m) ? -pos1 : pos1;
393 out[1] = pos0 > pos1 ? -out[1] : out[1];
394}
395
396static void decode_3p_track(int *out, int code, int m, int off) ///code: 3m+1 bits
397{
398 int half_2p = BIT_POS(code, 2*m - 1) << (m - 1);
399
400 decode_2p_track(out, BIT_STR(code, 0, 2*m - 1),
401 m - 1, off + half_2p);
402 decode_1p_track(out + 2, BIT_STR(code, 2*m, m + 1), m, off);
403}
404
405static void decode_4p_track(int *out, int code, int m, int off) ///code: 4m bits
406{
407 int half_4p, subhalf_2p;
408 int b_offset = 1 << (m - 1);
409
410 switch (BIT_STR(code, 4*m - 2, 2)) { /* case ID (2 bits) */
411 case 0: /* 0 pulses in A, 4 pulses in B or vice versa */
412 half_4p = BIT_POS(code, 4*m - 3) << (m - 1); // which has 4 pulses
413 subhalf_2p = BIT_POS(code, 2*m - 3) << (m - 2);
414
415 decode_2p_track(out, BIT_STR(code, 0, 2*m - 3),
416 m - 2, off + half_4p + subhalf_2p);
417 decode_2p_track(out + 2, BIT_STR(code, 2*m - 2, 2*m - 1),
418 m - 1, off + half_4p);
419 break;
420 case 1: /* 1 pulse in A, 3 pulses in B */
421 decode_1p_track(out, BIT_STR(code, 3*m - 2, m),
422 m - 1, off);
423 decode_3p_track(out + 1, BIT_STR(code, 0, 3*m - 2),
424 m - 1, off + b_offset);
425 break;
426 case 2: /* 2 pulses in each half */
427 decode_2p_track(out, BIT_STR(code, 2*m - 1, 2*m - 1),
428 m - 1, off);
429 decode_2p_track(out + 2, BIT_STR(code, 0, 2*m - 1),
430 m - 1, off + b_offset);
431 break;
432 case 3: /* 3 pulses in A, 1 pulse in B */
433 decode_3p_track(out, BIT_STR(code, m, 3*m - 2),
434 m - 1, off);
435 decode_1p_track(out + 3, BIT_STR(code, 0, m),
436 m - 1, off + b_offset);
437 break;
438 }
439}
440
441static void decode_5p_track(int *out, int code, int m, int off) ///code: 5m bits
442{
443 int half_3p = BIT_POS(code, 5*m - 1) << (m - 1);
444
445 decode_3p_track(out, BIT_STR(code, 2*m + 1, 3*m - 2),
446 m - 1, off + half_3p);
447
448 decode_2p_track(out + 3, BIT_STR(code, 0, 2*m + 1), m, off);
449}
450
451static void decode_6p_track(int *out, int code, int m, int off) ///code: 6m-2 bits
452{
453 int b_offset = 1 << (m - 1);
454 /* which half has more pulses in cases 0 to 2 */
455 int half_more = BIT_POS(code, 6*m - 5) << (m - 1);
456 int half_other = b_offset - half_more;
457
458 switch (BIT_STR(code, 6*m - 4, 2)) { /* case ID (2 bits) */
459 case 0: /* 0 pulses in A, 6 pulses in B or vice versa */
460 decode_1p_track(out, BIT_STR(code, 0, m),
461 m - 1, off + half_more);
462 decode_5p_track(out + 1, BIT_STR(code, m, 5*m - 5),
463 m - 1, off + half_more);
464 break;
465 case 1: /* 1 pulse in A, 5 pulses in B or vice versa */
466 decode_1p_track(out, BIT_STR(code, 0, m),
467 m - 1, off + half_other);
468 decode_5p_track(out + 1, BIT_STR(code, m, 5*m - 5),
469 m - 1, off + half_more);
470 break;
471 case 2: /* 2 pulses in A, 4 pulses in B or vice versa */
472 decode_2p_track(out, BIT_STR(code, 0, 2*m - 1),
473 m - 1, off + half_other);
474 decode_4p_track(out + 2, BIT_STR(code, 2*m - 1, 4*m - 4),
475 m - 1, off + half_more);
476 break;
477 case 3: /* 3 pulses in A, 3 pulses in B */
478 decode_3p_track(out, BIT_STR(code, 3*m - 2, 3*m - 2),
479 m - 1, off);
480 decode_3p_track(out + 3, BIT_STR(code, 0, 3*m - 2),
481 m - 1, off + b_offset);
482 break;
483 }
484}
485
486/**
487 * Decode the algebraic codebook index to pulse positions and signs,
488 * then construct the algebraic codebook vector.
489 *
490 * @param[out] fixed_vector Buffer for the fixed codebook excitation
491 * @param[in] pulse_hi MSBs part of the pulse index array (higher modes only)
492 * @param[in] pulse_lo LSBs part of the pulse index array
493 * @param[in] mode Mode of the current frame
494 */
495static void decode_fixed_vector(float *fixed_vector, const uint16_t *pulse_hi,
496 const uint16_t *pulse_lo, const enum Mode mode)
497{
498 /* sig_pos stores for each track the decoded pulse position indexes
499 * (1-based) multiplied by its corresponding amplitude (+1 or -1) */
500 int sig_pos[4][6];
501 int spacing = (mode == MODE_6k60) ? 2 : 4;
502 int i, j;
503
504 switch (mode) {
505 case MODE_6k60:
506 for (i = 0; i < 2; i++)
507 decode_1p_track(sig_pos[i], pulse_lo[i], 5, 1);
508 break;
509 case MODE_8k85:
510 for (i = 0; i < 4; i++)
511 decode_1p_track(sig_pos[i], pulse_lo[i], 4, 1);
512 break;
513 case MODE_12k65:
514 for (i = 0; i < 4; i++)
515 decode_2p_track(sig_pos[i], pulse_lo[i], 4, 1);
516 break;
517 case MODE_14k25:
518 for (i = 0; i < 2; i++)
519 decode_3p_track(sig_pos[i], pulse_lo[i], 4, 1);
520 for (i = 2; i < 4; i++)
521 decode_2p_track(sig_pos[i], pulse_lo[i], 4, 1);
522 break;
523 case MODE_15k85:
524 for (i = 0; i < 4; i++)
525 decode_3p_track(sig_pos[i], pulse_lo[i], 4, 1);
526 break;
527 case MODE_18k25:
528 for (i = 0; i < 4; i++)
529 decode_4p_track(sig_pos[i], (int) pulse_lo[i] +
530 ((int) pulse_hi[i] << 14), 4, 1);
531 break;
532 case MODE_19k85:
533 for (i = 0; i < 2; i++)
534 decode_5p_track(sig_pos[i], (int) pulse_lo[i] +
535 ((int) pulse_hi[i] << 10), 4, 1);
536 for (i = 2; i < 4; i++)
537 decode_4p_track(sig_pos[i], (int) pulse_lo[i] +
538 ((int) pulse_hi[i] << 14), 4, 1);
539 break;
540 case MODE_23k05:
541 case MODE_23k85:
542 for (i = 0; i < 4; i++)
543 decode_6p_track(sig_pos[i], (int) pulse_lo[i] +
544 ((int) pulse_hi[i] << 11), 4, 1);
545 break;
546 }
547
548 memset(fixed_vector, 0, sizeof(float) * AMRWB_SFR_SIZE);
549
550 for (i = 0; i < 4; i++)
551 for (j = 0; j < pulses_nb_per_mode_tr[mode][i]; j++) {
552 int pos = (FFABS(sig_pos[i][j]) - 1) * spacing + i;
553
554 fixed_vector[pos] += sig_pos[i][j] < 0 ? -1.0 : 1.0;
555 }
556}
557
558/**
559 * Decode pitch gain and fixed gain correction factor.
560 *
561 * @param[in] vq_gain Vector-quantized index for gains
562 * @param[in] mode Mode of the current frame
563 * @param[out] fixed_gain_factor Decoded fixed gain correction factor
564 * @param[out] pitch_gain Decoded pitch gain
565 */
566static void decode_gains(const uint8_t vq_gain, const enum Mode mode,
567 float *fixed_gain_factor, float *pitch_gain)
568{
569 const int16_t *gains = (mode <= MODE_8k85 ? qua_gain_6b[vq_gain] :
570 qua_gain_7b[vq_gain]);
571
572 *pitch_gain = gains[0] * (1.0f / (1 << 14));
573 *fixed_gain_factor = gains[1] * (1.0f / (1 << 11));
574}
575
576/**
577 * Apply pitch sharpening filters to the fixed codebook vector.
578 *
579 * @param[in] ctx The context
580 * @param[in,out] fixed_vector Fixed codebook excitation
581 */
582// XXX: Spec states this procedure should be applied when the pitch
583// lag is less than 64, but this checking seems absent in reference and AMR-NB
584static void pitch_sharpening(AMRWBContext *ctx, float *fixed_vector)
585{
586 int i;
587
588 /* Tilt part */
589 for (i = AMRWB_SFR_SIZE - 1; i != 0; i--)
590 fixed_vector[i] -= fixed_vector[i - 1] * ctx->tilt_coef;
591
592 /* Periodicity enhancement part */
593 for (i = ctx->pitch_lag_int; i < AMRWB_SFR_SIZE; i++)
594 fixed_vector[i] += fixed_vector[i - ctx->pitch_lag_int] * 0.85;
595}
596
597/**
598 * Calculate the voicing factor (-1.0 = unvoiced to 1.0 = voiced).
599 *
600 * @param[in] p_vector, f_vector Pitch and fixed excitation vectors
601 * @param[in] p_gain, f_gain Pitch and fixed gains
602 * @param[in] ctx The context
603 */
604// XXX: There is something wrong with the precision here! The magnitudes
605// of the energies are not correct. Please check the reference code carefully
606static float voice_factor(float *p_vector, float p_gain,
607 float *f_vector, float f_gain,
608 CELPMContext *ctx)
609{
610 double p_ener = (double) ctx->dot_productf(p_vector, p_vector,
611 AMRWB_SFR_SIZE) *
612 p_gain * p_gain;
613 double f_ener = (double) ctx->dot_productf(f_vector, f_vector,
614 AMRWB_SFR_SIZE) *
615 f_gain * f_gain;
616
617 return (p_ener - f_ener) / (p_ener + f_ener);
618}
619
620/**
621 * Reduce fixed vector sparseness by smoothing with one of three IR filters,
622 * also known as "adaptive phase dispersion".
623 *
624 * @param[in] ctx The context
625 * @param[in,out] fixed_vector Unfiltered fixed vector
626 * @param[out] buf Space for modified vector if necessary
627 *
628 * @return The potentially overwritten filtered fixed vector address
629 */
630static float *anti_sparseness(AMRWBContext *ctx,
631 float *fixed_vector, float *buf)
632{
633 int ir_filter_nr;
634
635 if (ctx->fr_cur_mode > MODE_8k85) // no filtering in higher modes
636 return fixed_vector;
637
638 if (ctx->pitch_gain[0] < 0.6) {
639 ir_filter_nr = 0; // strong filtering
640 } else if (ctx->pitch_gain[0] < 0.9) {
641 ir_filter_nr = 1; // medium filtering
642 } else
643 ir_filter_nr = 2; // no filtering
644
645 /* detect 'onset' */
646 if (ctx->fixed_gain[0] > 3.0 * ctx->fixed_gain[1]) {
647 if (ir_filter_nr < 2)
648 ir_filter_nr++;
649 } else {
650 int i, count = 0;
651
652 for (i = 0; i < 6; i++)
653 if (ctx->pitch_gain[i] < 0.6)
654 count++;
655
656 if (count > 2)
657 ir_filter_nr = 0;
658
659 if (ir_filter_nr > ctx->prev_ir_filter_nr + 1)
660 ir_filter_nr--;
661 }
662
663 /* update ir filter strength history */
664 ctx->prev_ir_filter_nr = ir_filter_nr;
665
666 ir_filter_nr += (ctx->fr_cur_mode == MODE_8k85);
667
668 if (ir_filter_nr < 2) {
669 int i;
670 const float *coef = ir_filters_lookup[ir_filter_nr];
671
672 /* Circular convolution code in the reference
673 * decoder was modified to avoid using one
674 * extra array. The filtered vector is given by:
675 *
676 * c2(n) = sum(i,0,len-1){ c(i) * coef( (n - i + len) % len ) }
677 */
678
679 memset(buf, 0, sizeof(float) * AMRWB_SFR_SIZE);
680 for (i = 0; i < AMRWB_SFR_SIZE; i++)
681 if (fixed_vector[i])
682 ff_celp_circ_addf(buf, buf, coef, i, fixed_vector[i],
683 AMRWB_SFR_SIZE);
684 fixed_vector = buf;
685 }
686
687 return fixed_vector;
688}
689
690/**
691 * Calculate a stability factor {teta} based on distance between
692 * current and past isf. A value of 1 shows maximum signal stability.
693 */
694static float stability_factor(const float *isf, const float *isf_past)
695{
696 int i;
697 float acc = 0.0;
698
699 for (i = 0; i < LP_ORDER - 1; i++)
700 acc += (isf[i] - isf_past[i]) * (isf[i] - isf_past[i]);
701
702 // XXX: This part is not so clear from the reference code
703 // the result is more accurate changing the "/ 256" to "* 512"
704 return FFMAX(0.0, 1.25 - acc * 0.8 * 512);
705}
706
707/**
708 * Apply a non-linear fixed gain smoothing in order to reduce
709 * fluctuation in the energy of excitation.
710 *
711 * @param[in] fixed_gain Unsmoothed fixed gain
712 * @param[in,out] prev_tr_gain Previous threshold gain (updated)
713 * @param[in] voice_fac Frame voicing factor
714 * @param[in] stab_fac Frame stability factor
715 *
716 * @return The smoothed gain
717 */
718static float noise_enhancer(float fixed_gain, float *prev_tr_gain,
719 float voice_fac, float stab_fac)
720{
721 float sm_fac = 0.5 * (1 - voice_fac) * stab_fac;
722 float g0;
723
724 // XXX: the following fixed-point constants used to in(de)crement
725 // gain by 1.5dB were taken from the reference code, maybe it could
726 // be simpler
727 if (fixed_gain < *prev_tr_gain) {
728 g0 = FFMIN(*prev_tr_gain, fixed_gain + fixed_gain *
729 (6226 * (1.0f / (1 << 15)))); // +1.5 dB
730 } else
731 g0 = FFMAX(*prev_tr_gain, fixed_gain *
732 (27536 * (1.0f / (1 << 15)))); // -1.5 dB
733
734 *prev_tr_gain = g0; // update next frame threshold
735
736 return sm_fac * g0 + (1 - sm_fac) * fixed_gain;
737}
738
739/**
740 * Filter the fixed_vector to emphasize the higher frequencies.
741 *
742 * @param[in,out] fixed_vector Fixed codebook vector
743 * @param[in] voice_fac Frame voicing factor
744 */
745static void pitch_enhancer(float *fixed_vector, float voice_fac)
746{
747 int i;
748 float cpe = 0.125 * (1 + voice_fac);
749 float last = fixed_vector[0]; // holds c(i - 1)
750
751 fixed_vector[0] -= cpe * fixed_vector[1];
752
753 for (i = 1; i < AMRWB_SFR_SIZE - 1; i++) {
754 float cur = fixed_vector[i];
755
756 fixed_vector[i] -= cpe * (last + fixed_vector[i + 1]);
757 last = cur;
758 }
759
760 fixed_vector[AMRWB_SFR_SIZE - 1] -= cpe * last;
761}
762
763/**
764 * Conduct 16th order linear predictive coding synthesis from excitation.
765 *
766 * @param[in] ctx Pointer to the AMRWBContext
767 * @param[in] lpc Pointer to the LPC coefficients
768 * @param[out] excitation Buffer for synthesis final excitation
769 * @param[in] fixed_gain Fixed codebook gain for synthesis
770 * @param[in] fixed_vector Algebraic codebook vector
771 * @param[in,out] samples Pointer to the output samples and memory
772 */
773static void synthesis(AMRWBContext *ctx, float *lpc, float *excitation,
774 float fixed_gain, const float *fixed_vector,
775 float *samples)
776{
777 ctx->acelpv_ctx.weighted_vector_sumf(excitation, ctx->pitch_vector, fixed_vector,
778 ctx->pitch_gain[0], fixed_gain, AMRWB_SFR_SIZE);
779
780 /* emphasize pitch vector contribution in low bitrate modes */
781 if (ctx->pitch_gain[0] > 0.5 && ctx->fr_cur_mode <= MODE_8k85) {
782 int i;
783 float energy = ctx->celpm_ctx.dot_productf(excitation, excitation,
784 AMRWB_SFR_SIZE);
785
786 // XXX: Weird part in both ref code and spec. A unknown parameter
787 // {beta} seems to be identical to the current pitch gain
788 float pitch_factor = 0.25 * ctx->pitch_gain[0] * ctx->pitch_gain[0];
789
790 for (i = 0; i < AMRWB_SFR_SIZE; i++)
791 excitation[i] += pitch_factor * ctx->pitch_vector[i];
792
793 ff_scale_vector_to_given_sum_of_squares(excitation, excitation,
794 energy, AMRWB_SFR_SIZE);
795 }
796
797 ctx->celpf_ctx.celp_lp_synthesis_filterf(samples, lpc, excitation,
798 AMRWB_SFR_SIZE, LP_ORDER);
799}
800
801/**
802 * Apply to synthesis a de-emphasis filter of the form:
803 * H(z) = 1 / (1 - m * z^-1)
804 *
805 * @param[out] out Output buffer
806 * @param[in] in Input samples array with in[-1]
807 * @param[in] m Filter coefficient
808 * @param[in,out] mem State from last filtering
809 */
810static void de_emphasis(float *out, float *in, float m, float mem[1])
811{
812 int i;
813
814 out[0] = in[0] + m * mem[0];
815
816 for (i = 1; i < AMRWB_SFR_SIZE; i++)
817 out[i] = in[i] + out[i - 1] * m;
818
819 mem[0] = out[AMRWB_SFR_SIZE - 1];
820}
821
822/**
823 * Upsample a signal by 5/4 ratio (from 12.8kHz to 16kHz) using
824 * a FIR interpolation filter. Uses past data from before *in address.
825 *
826 * @param[out] out Buffer for interpolated signal
827 * @param[in] in Current signal data (length 0.8*o_size)
828 * @param[in] o_size Output signal length
829 * @param[in] ctx The context
830 */
831static void upsample_5_4(float *out, const float *in, int o_size, CELPMContext *ctx)
832{
833 const float *in0 = in - UPS_FIR_SIZE + 1;
834 int i, j, k;
835 int int_part = 0, frac_part;
836
837 i = 0;
838 for (j = 0; j < o_size / 5; j++) {
839 out[i] = in[int_part];
840 frac_part = 4;
841 i++;
842
843 for (k = 1; k < 5; k++) {
844 out[i] = ctx->dot_productf(in0 + int_part,
845 upsample_fir[4 - frac_part],
846 UPS_MEM_SIZE);
847 int_part++;
848 frac_part--;
849 i++;
850 }
851 }
852}
853
854/**
855 * Calculate the high-band gain based on encoded index (23k85 mode) or
856 * on the low-band speech signal and the Voice Activity Detection flag.
857 *
858 * @param[in] ctx The context
859 * @param[in] synth LB speech synthesis at 12.8k
860 * @param[in] hb_idx Gain index for mode 23k85 only
861 * @param[in] vad VAD flag for the frame
862 */
863static float find_hb_gain(AMRWBContext *ctx, const float *synth,
864 uint16_t hb_idx, uint8_t vad)
865{
866 int wsp = (vad > 0);
867 float tilt;
868
869 if (ctx->fr_cur_mode == MODE_23k85)
870 return qua_hb_gain[hb_idx] * (1.0f / (1 << 14));
871
872 tilt = ctx->celpm_ctx.dot_productf(synth, synth + 1, AMRWB_SFR_SIZE - 1) /
873 ctx->celpm_ctx.dot_productf(synth, synth, AMRWB_SFR_SIZE);
874
875 /* return gain bounded by [0.1, 1.0] */
876 return av_clipf((1.0 - FFMAX(0.0, tilt)) * (1.25 - 0.25 * wsp), 0.1, 1.0);
877}
878
879/**
880 * Generate the high-band excitation with the same energy from the lower
881 * one and scaled by the given gain.
882 *
883 * @param[in] ctx The context
884 * @param[out] hb_exc Buffer for the excitation
885 * @param[in] synth_exc Low-band excitation used for synthesis
886 * @param[in] hb_gain Wanted excitation gain
887 */
888static void scaled_hb_excitation(AMRWBContext *ctx, float *hb_exc,
889 const float *synth_exc, float hb_gain)
890{
891 int i;
892 float energy = ctx->celpm_ctx.dot_productf(synth_exc, synth_exc,
893 AMRWB_SFR_SIZE);
894
895 /* Generate a white-noise excitation */
896 for (i = 0; i < AMRWB_SFR_SIZE_16k; i++)
897 hb_exc[i] = 32768.0 - (uint16_t) av_lfg_get(&ctx->prng);
898
899 ff_scale_vector_to_given_sum_of_squares(hb_exc, hb_exc,
900 energy * hb_gain * hb_gain,
901 AMRWB_SFR_SIZE_16k);
902}
903
904/**
905 * Calculate the auto-correlation for the ISF difference vector.
906 */
907static float auto_correlation(float *diff_isf, float mean, int lag)
908{
909 int i;
910 float sum = 0.0;
911
912 for (i = 7; i < LP_ORDER - 2; i++) {
913 float prod = (diff_isf[i] - mean) * (diff_isf[i - lag] - mean);
914 sum += prod * prod;
915 }
916 return sum;
917}
918
919/**
920 * Extrapolate a ISF vector to the 16kHz range (20th order LP)
921 * used at mode 6k60 LP filter for the high frequency band.
922 *
923 * @param[out] isf Buffer for extrapolated isf; contains LP_ORDER
924 * values on input
925 */
926static void extrapolate_isf(float isf[LP_ORDER_16k])
927{
928 float diff_isf[LP_ORDER - 2], diff_mean;
929 float corr_lag[3];
930 float est, scale;
931 int i, j, i_max_corr;
932
933 isf[LP_ORDER_16k - 1] = isf[LP_ORDER - 1];
934
935 /* Calculate the difference vector */
936 for (i = 0; i < LP_ORDER - 2; i++)
937 diff_isf[i] = isf[i + 1] - isf[i];
938
939 diff_mean = 0.0;
940 for (i = 2; i < LP_ORDER - 2; i++)
941 diff_mean += diff_isf[i] * (1.0f / (LP_ORDER - 4));
942
943 /* Find which is the maximum autocorrelation */
944 i_max_corr = 0;
945 for (i = 0; i < 3; i++) {
946 corr_lag[i] = auto_correlation(diff_isf, diff_mean, i + 2);
947
948 if (corr_lag[i] > corr_lag[i_max_corr])
949 i_max_corr = i;
950 }
951 i_max_corr++;
952
953 for (i = LP_ORDER - 1; i < LP_ORDER_16k - 1; i++)
954 isf[i] = isf[i - 1] + isf[i - 1 - i_max_corr]
955 - isf[i - 2 - i_max_corr];
956
957 /* Calculate an estimate for ISF(18) and scale ISF based on the error */
958 est = 7965 + (isf[2] - isf[3] - isf[4]) / 6.0;
959 scale = 0.5 * (FFMIN(est, 7600) - isf[LP_ORDER - 2]) /
960 (isf[LP_ORDER_16k - 2] - isf[LP_ORDER - 2]);
961
962 for (i = LP_ORDER - 1, j = 0; i < LP_ORDER_16k - 1; i++, j++)
963 diff_isf[j] = scale * (isf[i] - isf[i - 1]);
964
965 /* Stability insurance */
966 for (i = 1; i < LP_ORDER_16k - LP_ORDER; i++)
967 if (diff_isf[i] + diff_isf[i - 1] < 5.0) {
968 if (diff_isf[i] > diff_isf[i - 1]) {
969 diff_isf[i - 1] = 5.0 - diff_isf[i];
970 } else
971 diff_isf[i] = 5.0 - diff_isf[i - 1];
972 }
973
974 for (i = LP_ORDER - 1, j = 0; i < LP_ORDER_16k - 1; i++, j++)
975 isf[i] = isf[i - 1] + diff_isf[j] * (1.0f / (1 << 15));
976
977 /* Scale the ISF vector for 16000 Hz */
978 for (i = 0; i < LP_ORDER_16k - 1; i++)
979 isf[i] *= 0.8;
980}
981
982/**
983 * Spectral expand the LP coefficients using the equation:
984 * y[i] = x[i] * (gamma ** i)
985 *
986 * @param[out] out Output buffer (may use input array)
987 * @param[in] lpc LP coefficients array
988 * @param[in] gamma Weighting factor
989 * @param[in] size LP array size
990 */
991static void lpc_weighting(float *out, const float *lpc, float gamma, int size)
992{
993 int i;
994 float fac = gamma;
995
996 for (i = 0; i < size; i++) {
997 out[i] = lpc[i] * fac;
998 fac *= gamma;
999 }
1000}
1001
1002/**
1003 * Conduct 20th order linear predictive coding synthesis for the high
1004 * frequency band excitation at 16kHz.
1005 *
1006 * @param[in] ctx The context
1007 * @param[in] subframe Current subframe index (0 to 3)
1008 * @param[in,out] samples Pointer to the output speech samples
1009 * @param[in] exc Generated white-noise scaled excitation
1010 * @param[in] isf Current frame isf vector
1011 * @param[in] isf_past Past frame final isf vector
1012 */
1013static void hb_synthesis(AMRWBContext *ctx, int subframe, float *samples,
1014 const float *exc, const float *isf, const float *isf_past)
1015{
1016 float hb_lpc[LP_ORDER_16k];
1017 enum Mode mode = ctx->fr_cur_mode;
1018
1019 if (mode == MODE_6k60) {
1020 float e_isf[LP_ORDER_16k]; // ISF vector for extrapolation
1021 double e_isp[LP_ORDER_16k];
1022
1023 ctx->acelpv_ctx.weighted_vector_sumf(e_isf, isf_past, isf, isfp_inter[subframe],
1024 1.0 - isfp_inter[subframe], LP_ORDER);
1025
1026 extrapolate_isf(e_isf);
1027
1028 e_isf[LP_ORDER_16k - 1] *= 2.0;
1029 ff_acelp_lsf2lspd(e_isp, e_isf, LP_ORDER_16k);
1030 ff_amrwb_lsp2lpc(e_isp, hb_lpc, LP_ORDER_16k);
1031
1032 lpc_weighting(hb_lpc, hb_lpc, 0.9, LP_ORDER_16k);
1033 } else {
1034 lpc_weighting(hb_lpc, ctx->lp_coef[subframe], 0.6, LP_ORDER);
1035 }
1036
1037 ctx->celpf_ctx.celp_lp_synthesis_filterf(samples, hb_lpc, exc, AMRWB_SFR_SIZE_16k,
1038 (mode == MODE_6k60) ? LP_ORDER_16k : LP_ORDER);
1039}
1040
1041/**
1042 * Apply a 15th order filter to high-band samples.
1043 * The filter characteristic depends on the given coefficients.
1044 *
1045 * @param[out] out Buffer for filtered output
1046 * @param[in] fir_coef Filter coefficients
1047 * @param[in,out] mem State from last filtering (updated)
1048 * @param[in] in Input speech data (high-band)
1049 *
1050 * @remark It is safe to pass the same array in in and out parameters
1051 */
1052
1053#ifndef hb_fir_filter
1054static void hb_fir_filter(float *out, const float fir_coef[HB_FIR_SIZE + 1],
1055 float mem[HB_FIR_SIZE], const float *in)
1056{
1057 int i, j;
1058 float data[AMRWB_SFR_SIZE_16k + HB_FIR_SIZE]; // past and current samples
1059
1060 memcpy(data, mem, HB_FIR_SIZE * sizeof(float));
1061 memcpy(data + HB_FIR_SIZE, in, AMRWB_SFR_SIZE_16k * sizeof(float));
1062
1063 for (i = 0; i < AMRWB_SFR_SIZE_16k; i++) {
1064 out[i] = 0.0;
1065 for (j = 0; j <= HB_FIR_SIZE; j++)
1066 out[i] += data[i + j] * fir_coef[j];
1067 }
1068
1069 memcpy(mem, data + AMRWB_SFR_SIZE_16k, HB_FIR_SIZE * sizeof(float));
1070}
1071#endif /* hb_fir_filter */
1072
1073/**
1074 * Update context state before the next subframe.
1075 */
1076static void update_sub_state(AMRWBContext *ctx)
1077{
1078 memmove(&ctx->excitation_buf[0], &ctx->excitation_buf[AMRWB_SFR_SIZE],
1079 (AMRWB_P_DELAY_MAX + LP_ORDER + 1) * sizeof(float));
1080
1081 memmove(&ctx->pitch_gain[1], &ctx->pitch_gain[0], 5 * sizeof(float));
1082 memmove(&ctx->fixed_gain[1], &ctx->fixed_gain[0], 1 * sizeof(float));
1083
1084 memmove(&ctx->samples_az[0], &ctx->samples_az[AMRWB_SFR_SIZE],
1085 LP_ORDER * sizeof(float));
1086 memmove(&ctx->samples_up[0], &ctx->samples_up[AMRWB_SFR_SIZE],
1087 UPS_MEM_SIZE * sizeof(float));
1088 memmove(&ctx->samples_hb[0], &ctx->samples_hb[AMRWB_SFR_SIZE_16k],
1089 LP_ORDER_16k * sizeof(float));
1090}
1091
1092static int amrwb_decode_frame(AVCodecContext *avctx, void *data,
1093 int *got_frame_ptr, AVPacket *avpkt)
1094{
1095 AMRWBContext *ctx = avctx->priv_data;
1096 AVFrame *frame = data;
1097 AMRWBFrame *cf = &ctx->frame;
1098 const uint8_t *buf = avpkt->data;
1099 int buf_size = avpkt->size;
1100 int expected_fr_size, header_size;
1101 float *buf_out;
1102 float spare_vector[AMRWB_SFR_SIZE]; // extra stack space to hold result from anti-sparseness processing
1103 float fixed_gain_factor; // fixed gain correction factor (gamma)
1104 float *synth_fixed_vector; // pointer to the fixed vector that synthesis should use
1105 float synth_fixed_gain; // the fixed gain that synthesis should use
1106 float voice_fac, stab_fac; // parameters used for gain smoothing
1107 float synth_exc[AMRWB_SFR_SIZE]; // post-processed excitation for synthesis
1108 float hb_exc[AMRWB_SFR_SIZE_16k]; // excitation for the high frequency band
1109 float hb_samples[AMRWB_SFR_SIZE_16k]; // filtered high-band samples from synthesis
1110 float hb_gain;
1111 int sub, i, ret;
1112
1113 /* get output buffer */
1114 frame->nb_samples = 4 * AMRWB_SFR_SIZE_16k;
1115 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
1116 return ret;
1117 buf_out = (float *)frame->data[0];
1118
1119 header_size = decode_mime_header(ctx, buf);
1120 if (ctx->fr_cur_mode > MODE_SID) {
1121 av_log(avctx, AV_LOG_ERROR,
1122 "Invalid mode %d\n", ctx->fr_cur_mode);
1123 return AVERROR_INVALIDDATA;
1124 }
1125 expected_fr_size = ((cf_sizes_wb[ctx->fr_cur_mode] + 7) >> 3) + 1;
1126
1127 if (buf_size < expected_fr_size) {
1128 av_log(avctx, AV_LOG_ERROR,
1129 "Frame too small (%d bytes). Truncated file?\n", buf_size);
1130 *got_frame_ptr = 0;
1131 return AVERROR_INVALIDDATA;
1132 }
1133
1134 if (!ctx->fr_quality || ctx->fr_cur_mode > MODE_SID)
1135 av_log(avctx, AV_LOG_ERROR, "Encountered a bad or corrupted frame\n");
1136
1137 if (ctx->fr_cur_mode == MODE_SID) { /* Comfort noise frame */
1138 avpriv_request_sample(avctx, "SID mode");
1139 return AVERROR_PATCHWELCOME;
1140 }
1141
1142 ff_amr_bit_reorder((uint16_t *) &ctx->frame, sizeof(AMRWBFrame),
1143 buf + header_size, amr_bit_orderings_by_mode[ctx->fr_cur_mode]);
1144
1145 /* Decode the quantized ISF vector */
1146 if (ctx->fr_cur_mode == MODE_6k60) {
1147 decode_isf_indices_36b(cf->isp_id, ctx->isf_cur);
1148 } else {
1149 decode_isf_indices_46b(cf->isp_id, ctx->isf_cur);
1150 }
1151
1152 isf_add_mean_and_past(ctx->isf_cur, ctx->isf_q_past);
1153 ff_set_min_dist_lsf(ctx->isf_cur, MIN_ISF_SPACING, LP_ORDER - 1);
1154
1155 stab_fac = stability_factor(ctx->isf_cur, ctx->isf_past_final);
1156
1157 ctx->isf_cur[LP_ORDER - 1] *= 2.0;
1158 ff_acelp_lsf2lspd(ctx->isp[3], ctx->isf_cur, LP_ORDER);
1159
1160 /* Generate a ISP vector for each subframe */
1161 if (ctx->first_frame) {
1162 ctx->first_frame = 0;
1163 memcpy(ctx->isp_sub4_past, ctx->isp[3], LP_ORDER * sizeof(double));
1164 }
1165 interpolate_isp(ctx->isp, ctx->isp_sub4_past);
1166
1167 for (sub = 0; sub < 4; sub++)
1168 ff_amrwb_lsp2lpc(ctx->isp[sub], ctx->lp_coef[sub], LP_ORDER);
1169
1170 for (sub = 0; sub < 4; sub++) {
1171 const AMRWBSubFrame *cur_subframe = &cf->subframe[sub];
1172 float *sub_buf = buf_out + sub * AMRWB_SFR_SIZE_16k;
1173
1174 /* Decode adaptive codebook (pitch vector) */
1175 decode_pitch_vector(ctx, cur_subframe, sub);
1176 /* Decode innovative codebook (fixed vector) */
1177 decode_fixed_vector(ctx->fixed_vector, cur_subframe->pul_ih,
1178 cur_subframe->pul_il, ctx->fr_cur_mode);
1179
1180 pitch_sharpening(ctx, ctx->fixed_vector);
1181
1182 decode_gains(cur_subframe->vq_gain, ctx->fr_cur_mode,
1183 &fixed_gain_factor, &ctx->pitch_gain[0]);
1184
1185 ctx->fixed_gain[0] =
1186 ff_amr_set_fixed_gain(fixed_gain_factor,
1187 ctx->celpm_ctx.dot_productf(ctx->fixed_vector,
1188 ctx->fixed_vector,
1189 AMRWB_SFR_SIZE) /
1190 AMRWB_SFR_SIZE,
1191 ctx->prediction_error,
1192 ENERGY_MEAN, energy_pred_fac);
1193
1194 /* Calculate voice factor and store tilt for next subframe */
1195 voice_fac = voice_factor(ctx->pitch_vector, ctx->pitch_gain[0],
1196 ctx->fixed_vector, ctx->fixed_gain[0],
1197 &ctx->celpm_ctx);
1198 ctx->tilt_coef = voice_fac * 0.25 + 0.25;
1199
1200 /* Construct current excitation */
1201 for (i = 0; i < AMRWB_SFR_SIZE; i++) {
1202 ctx->excitation[i] *= ctx->pitch_gain[0];
1203 ctx->excitation[i] += ctx->fixed_gain[0] * ctx->fixed_vector[i];
1204 ctx->excitation[i] = truncf(ctx->excitation[i]);
1205 }
1206
1207 /* Post-processing of excitation elements */
1208 synth_fixed_gain = noise_enhancer(ctx->fixed_gain[0], &ctx->prev_tr_gain,
1209 voice_fac, stab_fac);
1210
1211 synth_fixed_vector = anti_sparseness(ctx, ctx->fixed_vector,
1212 spare_vector);
1213
1214 pitch_enhancer(synth_fixed_vector, voice_fac);
1215
1216 synthesis(ctx, ctx->lp_coef[sub], synth_exc, synth_fixed_gain,
1217 synth_fixed_vector, &ctx->samples_az[LP_ORDER]);
1218
1219 /* Synthesis speech post-processing */
1220 de_emphasis(&ctx->samples_up[UPS_MEM_SIZE],
1221 &ctx->samples_az[LP_ORDER], PREEMPH_FAC, ctx->demph_mem);
1222
1223 ctx->acelpf_ctx.acelp_apply_order_2_transfer_function(&ctx->samples_up[UPS_MEM_SIZE],
1224 &ctx->samples_up[UPS_MEM_SIZE], hpf_zeros, hpf_31_poles,
1225 hpf_31_gain, ctx->hpf_31_mem, AMRWB_SFR_SIZE);
1226
1227 upsample_5_4(sub_buf, &ctx->samples_up[UPS_FIR_SIZE],
1228 AMRWB_SFR_SIZE_16k, &ctx->celpm_ctx);
1229
1230 /* High frequency band (6.4 - 7.0 kHz) generation part */
1231 ctx->acelpf_ctx.acelp_apply_order_2_transfer_function(hb_samples,
1232 &ctx->samples_up[UPS_MEM_SIZE], hpf_zeros, hpf_400_poles,
1233 hpf_400_gain, ctx->hpf_400_mem, AMRWB_SFR_SIZE);
1234
1235 hb_gain = find_hb_gain(ctx, hb_samples,
1236 cur_subframe->hb_gain, cf->vad);
1237
1238 scaled_hb_excitation(ctx, hb_exc, synth_exc, hb_gain);
1239
1240 hb_synthesis(ctx, sub, &ctx->samples_hb[LP_ORDER_16k],
1241 hb_exc, ctx->isf_cur, ctx->isf_past_final);
1242
1243 /* High-band post-processing filters */
1244 hb_fir_filter(hb_samples, bpf_6_7_coef, ctx->bpf_6_7_mem,
1245 &ctx->samples_hb[LP_ORDER_16k]);
1246
1247 if (ctx->fr_cur_mode == MODE_23k85)
1248 hb_fir_filter(hb_samples, lpf_7_coef, ctx->lpf_7_mem,
1249 hb_samples);
1250
1251 /* Add the low and high frequency bands */
1252 for (i = 0; i < AMRWB_SFR_SIZE_16k; i++)
1253 sub_buf[i] = (sub_buf[i] + hb_samples[i]) * (1.0f / (1 << 15));
1254
1255 /* Update buffers and history */
1256 update_sub_state(ctx);
1257 }
1258
1259 /* update state for next frame */
1260 memcpy(ctx->isp_sub4_past, ctx->isp[3], LP_ORDER * sizeof(ctx->isp[3][0]));
1261 memcpy(ctx->isf_past_final, ctx->isf_cur, LP_ORDER * sizeof(float));
1262
1263 *got_frame_ptr = 1;
1264
1265 return expected_fr_size;
1266}
1267
1268AVCodec ff_amrwb_decoder = {
1269 .name = "amrwb",
1270 .long_name = NULL_IF_CONFIG_SMALL("AMR-WB (Adaptive Multi-Rate WideBand)"),
1271 .type = AVMEDIA_TYPE_AUDIO,
1272 .id = AV_CODEC_ID_AMR_WB,
1273 .priv_data_size = sizeof(AMRWBContext),
1274 .init = amrwb_decode_init,
1275 .decode = amrwb_decode_frame,
1276 .capabilities = CODEC_CAP_DR1,
1277 .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT,
1278 AV_SAMPLE_FMT_NONE },
1279};