FFmpeg coverage


Directory: ../../../ffmpeg/
File: src/libavcodec/amrnbdec.c
Date: 2022-07-07 01:21:54
Exec Total Coverage
Lines: 368 388 94.8%
Branches: 185 204 90.7%

Line Branch Exec Source
1 /*
2 * AMR narrowband decoder
3 * Copyright (c) 2006-2007 Robert Swain
4 * Copyright (c) 2009 Colin McQuillan
5 *
6 * This file is part of FFmpeg.
7 *
8 * FFmpeg is free software; you can redistribute it and/or
9 * modify it under the terms of the GNU Lesser General Public
10 * License as published by the Free Software Foundation; either
11 * version 2.1 of the License, or (at your option) any later version.
12 *
13 * FFmpeg is distributed in the hope that it will be useful,
14 * but WITHOUT ANY WARRANTY; without even the implied warranty of
15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
16 * Lesser General Public License for more details.
17 *
18 * You should have received a copy of the GNU Lesser General Public
19 * License along with FFmpeg; if not, write to the Free Software
20 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
21 */
22
23
24 /**
25 * @file
26 * AMR narrowband decoder
27 *
28 * This decoder uses floats for simplicity and so is not bit-exact. One
29 * difference is that differences in phase can accumulate. The test sequences
30 * in 3GPP TS 26.074 can still be useful.
31 *
32 * - Comparing this file's output to the output of the ref decoder gives a
33 * PSNR of 30 to 80. Plotting the output samples shows a difference in
34 * phase in some areas.
35 *
36 * - Comparing both decoders against their input, this decoder gives a similar
37 * PSNR. If the test sequence homing frames are removed (this decoder does
38 * not detect them), the PSNR is at least as good as the reference on 140
39 * out of 169 tests.
40 */
41
42
43 #include <string.h>
44 #include <math.h>
45
46 #include "libavutil/channel_layout.h"
47 #include "libavutil/float_dsp.h"
48 #include "avcodec.h"
49 #include "libavutil/common.h"
50 #include "libavutil/avassert.h"
51 #include "celp_math.h"
52 #include "celp_filters.h"
53 #include "acelp_filters.h"
54 #include "acelp_vectors.h"
55 #include "acelp_pitch_delay.h"
56 #include "lsp.h"
57 #include "amr.h"
58 #include "codec_internal.h"
59 #include "internal.h"
60
61 #include "amrnbdata.h"
62
63 #define AMR_BLOCK_SIZE 160 ///< samples per frame
64 #define AMR_SAMPLE_BOUND 32768.0 ///< threshold for synthesis overflow
65
66 /**
67 * Scale from constructed speech to [-1,1]
68 *
69 * AMR is designed to produce 16-bit PCM samples (3GPP TS 26.090 4.2) but
70 * upscales by two (section 6.2.2).
71 *
72 * Fundamentally, this scale is determined by energy_mean through
73 * the fixed vector contribution to the excitation vector.
74 */
75 #define AMR_SAMPLE_SCALE (2.0 / 32768.0)
76
77 /** Prediction factor for 12.2kbit/s mode */
78 #define PRED_FAC_MODE_12k2 0.65
79
80 #define LSF_R_FAC (8000.0 / 32768.0) ///< LSF residual tables to Hertz
81 #define MIN_LSF_SPACING (50.0488 / 8000.0) ///< Ensures stability of LPC filter
82 #define PITCH_LAG_MIN_MODE_12k2 18 ///< Lower bound on decoded lag search in 12.2kbit/s mode
83
84 /** Initial energy in dB. Also used for bad frames (unimplemented). */
85 #define MIN_ENERGY -14.0
86
87 /** Maximum sharpening factor
88 *
89 * The specification says 0.8, which should be 13107, but the reference C code
90 * uses 13017 instead. (Amusingly the same applies to SHARP_MAX in g729dec.c.)
91 */
92 #define SHARP_MAX 0.79449462890625
93
94 /** Number of impulse response coefficients used for tilt factor */
95 #define AMR_TILT_RESPONSE 22
96 /** Tilt factor = 1st reflection coefficient * gamma_t */
97 #define AMR_TILT_GAMMA_T 0.8
98 /** Adaptive gain control factor used in post-filter */
99 #define AMR_AGC_ALPHA 0.9
100
101 typedef struct AMRContext {
102 AMRNBFrame frame; ///< decoded AMR parameters (lsf coefficients, codebook indexes, etc)
103 uint8_t bad_frame_indicator; ///< bad frame ? 1 : 0
104 enum Mode cur_frame_mode;
105
106 int16_t prev_lsf_r[LP_FILTER_ORDER]; ///< residual LSF vector from previous subframe
107 double lsp[4][LP_FILTER_ORDER]; ///< lsp vectors from current frame
108 double prev_lsp_sub4[LP_FILTER_ORDER]; ///< lsp vector for the 4th subframe of the previous frame
109
110 float lsf_q[4][LP_FILTER_ORDER]; ///< Interpolated LSF vector for fixed gain smoothing
111 float lsf_avg[LP_FILTER_ORDER]; ///< vector of averaged lsf vector
112
113 float lpc[4][LP_FILTER_ORDER]; ///< lpc coefficient vectors for 4 subframes
114
115 uint8_t pitch_lag_int; ///< integer part of pitch lag from current subframe
116
117 float excitation_buf[PITCH_DELAY_MAX + LP_FILTER_ORDER + 1 + AMR_SUBFRAME_SIZE]; ///< current excitation and all necessary excitation history
118 float *excitation; ///< pointer to the current excitation vector in excitation_buf
119
120 float pitch_vector[AMR_SUBFRAME_SIZE]; ///< adaptive code book (pitch) vector
121 float fixed_vector[AMR_SUBFRAME_SIZE]; ///< algebraic codebook (fixed) vector (must be kept zero between frames)
122
123 float prediction_error[4]; ///< quantified prediction errors {20log10(^gamma_gc)} for previous four subframes
124 float pitch_gain[5]; ///< quantified pitch gains for the current and previous four subframes
125 float fixed_gain[5]; ///< quantified fixed gains for the current and previous four subframes
126
127 float beta; ///< previous pitch_gain, bounded by [0.0,SHARP_MAX]
128 uint8_t diff_count; ///< the number of subframes for which diff has been above 0.65
129 uint8_t hang_count; ///< the number of subframes since a hangover period started
130
131 float prev_sparse_fixed_gain; ///< previous fixed gain; used by anti-sparseness processing to determine "onset"
132 uint8_t prev_ir_filter_nr; ///< previous impulse response filter "impNr": 0 - strong, 1 - medium, 2 - none
133 uint8_t ir_filter_onset; ///< flag for impulse response filter strength
134
135 float postfilter_mem[10]; ///< previous intermediate values in the formant filter
136 float tilt_mem; ///< previous input to tilt compensation filter
137 float postfilter_agc; ///< previous factor used for adaptive gain control
138 float high_pass_mem[2]; ///< previous intermediate values in the high-pass filter
139
140 float samples_in[LP_FILTER_ORDER + AMR_SUBFRAME_SIZE]; ///< floating point samples
141
142 ACELPFContext acelpf_ctx; ///< context for filters for ACELP-based codecs
143 ACELPVContext acelpv_ctx; ///< context for vector operations for ACELP-based codecs
144 CELPFContext celpf_ctx; ///< context for filters for CELP-based codecs
145 CELPMContext celpm_ctx; ///< context for fixed point math operations
146
147 } AMRContext;
148
149 typedef struct AMRChannelsContext {
150 AMRContext ch[2];
151 } AMRChannelsContext;
152
153 /** Double version of ff_weighted_vector_sumf() */
154 570 static void weighted_vector_sumd(double *out, const double *in_a,
155 const double *in_b, double weight_coeff_a,
156 double weight_coeff_b, int length)
157 {
158 int i;
159
160
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161 5700 out[i] = weight_coeff_a * in_a[i]
162 5700 + weight_coeff_b * in_b[i];
163 570 }
164
165 20 static av_cold int amrnb_decode_init(AVCodecContext *avctx)
166 {
167 20 AMRChannelsContext *s = avctx->priv_data;
168 int i;
169
170
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20 if (avctx->ch_layout.nb_channels > 2) {
171 avpriv_report_missing_feature(avctx, ">2 channel AMR");
172 return AVERROR_PATCHWELCOME;
173 }
174
175
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20 if (!avctx->ch_layout.nb_channels) {
176 av_channel_layout_uninit(&avctx->ch_layout);
177 avctx->ch_layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_MONO;
178 }
179
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20 if (!avctx->sample_rate)
180 avctx->sample_rate = 8000;
181 20 avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
182
183
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40 for (int ch = 0; ch < avctx->ch_layout.nb_channels; ch++) {
184 20 AMRContext *p = &s->ch[ch];
185 // p->excitation always points to the same position in p->excitation_buf
186 20 p->excitation = &p->excitation_buf[PITCH_DELAY_MAX + LP_FILTER_ORDER + 1];
187
188
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220 for (i = 0; i < LP_FILTER_ORDER; i++) {
189 200 p->prev_lsp_sub4[i] = lsp_sub4_init[i] * 1000 / (float)(1 << 15);
190 200 p->lsf_avg[i] = p->lsf_q[3][i] = lsp_avg_init[i] / (float)(1 << 15);
191 }
192
193
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100 for (i = 0; i < 4; i++)
194 80 p->prediction_error[i] = MIN_ENERGY;
195
196 20 ff_acelp_filter_init(&p->acelpf_ctx);
197 20 ff_acelp_vectors_init(&p->acelpv_ctx);
198 20 ff_celp_filter_init(&p->celpf_ctx);
199 20 ff_celp_math_init(&p->celpm_ctx);
200 }
201
202 20 return 0;
203 }
204
205
206 /**
207 * Unpack an RFC4867 speech frame into the AMR frame mode and parameters.
208 *
209 * The order of speech bits is specified by 3GPP TS 26.101.
210 *
211 * @param p the context
212 * @param buf pointer to the input buffer
213 * @param buf_size size of the input buffer
214 *
215 * @return the frame mode
216 */
217 2284 static enum Mode unpack_bitstream(AMRContext *p, const uint8_t *buf,
218 int buf_size)
219 {
220 enum Mode mode;
221
222 // Decode the first octet.
223 2284 mode = buf[0] >> 3 & 0x0F; // frame type
224 2284 p->bad_frame_indicator = (buf[0] & 0x4) != 0x4; // quality bit
225
226
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2284 if (mode >= N_MODES || buf_size < frame_sizes_nb[mode] + 1) {
227 return NO_DATA;
228 }
229
230
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2284 if (mode < MODE_DTX)
231 2284 ff_amr_bit_reorder((uint16_t *) &p->frame, sizeof(AMRNBFrame), buf + 1,
232 2284 amr_unpacking_bitmaps_per_mode[mode]);
233
234 2284 return mode;
235 }
236
237
238 /// @name AMR pitch LPC coefficient decoding functions
239 /// @{
240
241 /**
242 * Interpolate the LSF vector (used for fixed gain smoothing).
243 * The interpolation is done over all four subframes even in MODE_12k2.
244 *
245 * @param[in] ctx The Context
246 * @param[in,out] lsf_q LSFs in [0,1] for each subframe
247 * @param[in] lsf_new New LSFs in [0,1] for subframe 4
248 */
249 2284 static void interpolate_lsf(ACELPVContext *ctx, float lsf_q[4][LP_FILTER_ORDER], float *lsf_new)
250 {
251 int i;
252
253
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11420 for (i = 0; i < 4; i++)
254 9136 ctx->weighted_vector_sumf(lsf_q[i], lsf_q[3], lsf_new,
255 9136 0.25 * (3 - i), 0.25 * (i + 1),
256 LP_FILTER_ORDER);
257 2284 }
258
259 /**
260 * Decode a set of 5 split-matrix quantized lsf indexes into an lsp vector.
261 *
262 * @param p the context
263 * @param lsp output LSP vector
264 * @param lsf_no_r LSF vector without the residual vector added
265 * @param lsf_quantizer pointers to LSF dictionary tables
266 * @param quantizer_offset offset in tables
267 * @param sign for the 3 dictionary table
268 * @param update store data for computing the next frame's LSFs
269 */
270 570 static void lsf2lsp_for_mode12k2(AMRContext *p, double lsp[LP_FILTER_ORDER],
271 const float lsf_no_r[LP_FILTER_ORDER],
272 const int16_t *lsf_quantizer[5],
273 const int quantizer_offset,
274 const int sign, const int update)
275 {
276 int16_t lsf_r[LP_FILTER_ORDER]; // residual LSF vector
277 float lsf_q[LP_FILTER_ORDER]; // quantified LSF vector
278 int i;
279
280
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3420 for (i = 0; i < LP_FILTER_ORDER >> 1; i++)
281 2850 memcpy(&lsf_r[i << 1], &lsf_quantizer[i][quantizer_offset],
282 2 * sizeof(*lsf_r));
283
284
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570 if (sign) {
285 266 lsf_r[4] *= -1;
286 266 lsf_r[5] *= -1;
287 }
288
289
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570 if (update)
290 285 memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(*lsf_r));
291
292
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6270 for (i = 0; i < LP_FILTER_ORDER; i++)
293 5700 lsf_q[i] = lsf_r[i] * (LSF_R_FAC / 8000.0) + lsf_no_r[i] * (1.0 / 8000.0);
294
295 570 ff_set_min_dist_lsf(lsf_q, MIN_LSF_SPACING, LP_FILTER_ORDER);
296
297
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570 if (update)
298 285 interpolate_lsf(&p->acelpv_ctx, p->lsf_q, lsf_q);
299
300 570 ff_acelp_lsf2lspd(lsp, lsf_q, LP_FILTER_ORDER);
301 570 }
302
303 /**
304 * Decode a set of 5 split-matrix quantized lsf indexes into 2 lsp vectors.
305 *
306 * @param p pointer to the AMRContext
307 */
308 285 static void lsf2lsp_5(AMRContext *p)
309 {
310 285 const uint16_t *lsf_param = p->frame.lsf;
311 float lsf_no_r[LP_FILTER_ORDER]; // LSFs without the residual vector
312 const int16_t *lsf_quantizer[5];
313 int i;
314
315 285 lsf_quantizer[0] = lsf_5_1[lsf_param[0]];
316 285 lsf_quantizer[1] = lsf_5_2[lsf_param[1]];
317 285 lsf_quantizer[2] = lsf_5_3[lsf_param[2] >> 1];
318 285 lsf_quantizer[3] = lsf_5_4[lsf_param[3]];
319 285 lsf_quantizer[4] = lsf_5_5[lsf_param[4]];
320
321
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3135 for (i = 0; i < LP_FILTER_ORDER; i++)
322 2850 lsf_no_r[i] = p->prev_lsf_r[i] * LSF_R_FAC * PRED_FAC_MODE_12k2 + lsf_5_mean[i];
323
324 285 lsf2lsp_for_mode12k2(p, p->lsp[1], lsf_no_r, lsf_quantizer, 0, lsf_param[2] & 1, 0);
325 285 lsf2lsp_for_mode12k2(p, p->lsp[3], lsf_no_r, lsf_quantizer, 2, lsf_param[2] & 1, 1);
326
327 // interpolate LSP vectors at subframes 1 and 3
328 285 weighted_vector_sumd(p->lsp[0], p->prev_lsp_sub4, p->lsp[1], 0.5, 0.5, LP_FILTER_ORDER);
329 285 weighted_vector_sumd(p->lsp[2], p->lsp[1] , p->lsp[3], 0.5, 0.5, LP_FILTER_ORDER);
330 285 }
331
332 /**
333 * Decode a set of 3 split-matrix quantized lsf indexes into an lsp vector.
334 *
335 * @param p pointer to the AMRContext
336 */
337 1999 static void lsf2lsp_3(AMRContext *p)
338 {
339 1999 const uint16_t *lsf_param = p->frame.lsf;
340 int16_t lsf_r[LP_FILTER_ORDER]; // residual LSF vector
341 float lsf_q[LP_FILTER_ORDER]; // quantified LSF vector
342 const int16_t *lsf_quantizer;
343 int i, j;
344
345
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1999 lsf_quantizer = (p->cur_frame_mode == MODE_7k95 ? lsf_3_1_MODE_7k95 : lsf_3_1)[lsf_param[0]];
346 1999 memcpy(lsf_r, lsf_quantizer, 3 * sizeof(*lsf_r));
347
348 1999 lsf_quantizer = lsf_3_2[lsf_param[1] << (p->cur_frame_mode <= MODE_5k15)];
349 1999 memcpy(lsf_r + 3, lsf_quantizer, 3 * sizeof(*lsf_r));
350
351
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1999 lsf_quantizer = (p->cur_frame_mode <= MODE_5k15 ? lsf_3_3_MODE_5k15 : lsf_3_3)[lsf_param[2]];
352 1999 memcpy(lsf_r + 6, lsf_quantizer, 4 * sizeof(*lsf_r));
353
354 // calculate mean-removed LSF vector and add mean
355
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21989 for (i = 0; i < LP_FILTER_ORDER; i++)
356 19990 lsf_q[i] = (lsf_r[i] + p->prev_lsf_r[i] * pred_fac[i]) * (LSF_R_FAC / 8000.0) + lsf_3_mean[i] * (1.0 / 8000.0);
357
358 1999 ff_set_min_dist_lsf(lsf_q, MIN_LSF_SPACING, LP_FILTER_ORDER);
359
360 // store data for computing the next frame's LSFs
361 1999 interpolate_lsf(&p->acelpv_ctx, p->lsf_q, lsf_q);
362 1999 memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(*lsf_r));
363
364 1999 ff_acelp_lsf2lspd(p->lsp[3], lsf_q, LP_FILTER_ORDER);
365
366 // interpolate LSP vectors at subframes 1, 2 and 3
367
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7996 for (i = 1; i <= 3; i++)
368
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65967 for(j = 0; j < LP_FILTER_ORDER; j++)
369 59970 p->lsp[i-1][j] = p->prev_lsp_sub4[j] +
370 59970 (p->lsp[3][j] - p->prev_lsp_sub4[j]) * 0.25 * i;
371 1999 }
372
373 /// @}
374
375
376 /// @name AMR pitch vector decoding functions
377 /// @{
378
379 /**
380 * Like ff_decode_pitch_lag(), but with 1/6 resolution
381 */
382 1140 static void decode_pitch_lag_1_6(int *lag_int, int *lag_frac, int pitch_index,
383 const int prev_lag_int, const int subframe)
384 {
385
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1140 if (subframe == 0 || subframe == 2) {
386
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570 if (pitch_index < 463) {
387 459 *lag_int = (pitch_index + 107) * 10923 >> 16;
388 459 *lag_frac = pitch_index - *lag_int * 6 + 105;
389 } else {
390 111 *lag_int = pitch_index - 368;
391 111 *lag_frac = 0;
392 }
393 } else {
394 570 *lag_int = ((pitch_index + 5) * 10923 >> 16) - 1;
395 570 *lag_frac = pitch_index - *lag_int * 6 - 3;
396 570 *lag_int += av_clip(prev_lag_int - 5, PITCH_LAG_MIN_MODE_12k2,
397 PITCH_DELAY_MAX - 9);
398 }
399 1140 }
400
401 9136 static void decode_pitch_vector(AMRContext *p,
402 const AMRNBSubframe *amr_subframe,
403 const int subframe)
404 {
405 int pitch_lag_int, pitch_lag_frac;
406 9136 enum Mode mode = p->cur_frame_mode;
407
408
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9136 if (p->cur_frame_mode == MODE_12k2) {
409 1140 decode_pitch_lag_1_6(&pitch_lag_int, &pitch_lag_frac,
410 1140 amr_subframe->p_lag, p->pitch_lag_int,
411 subframe);
412 } else {
413
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11424 ff_decode_pitch_lag(&pitch_lag_int, &pitch_lag_frac,
414 7996 amr_subframe->p_lag,
415 7996 p->pitch_lag_int, subframe,
416 mode != MODE_4k75 && mode != MODE_5k15,
417
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3428 mode <= MODE_6k7 ? 4 : (mode == MODE_7k95 ? 5 : 6));
418 7996 pitch_lag_frac *= 2;
419 }
420
421 9136 p->pitch_lag_int = pitch_lag_int; // store previous lag in a uint8_t
422
423 9136 pitch_lag_int += pitch_lag_frac > 0;
424
425 /* Calculate the pitch vector by interpolating the past excitation at the
426 pitch lag using a b60 hamming windowed sinc function. */
427 18272 p->acelpf_ctx.acelp_interpolatef(p->excitation,
428 9136 p->excitation + 1 - pitch_lag_int,
429 ff_b60_sinc, 6,
430
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9136 pitch_lag_frac + 6 - 6*(pitch_lag_frac > 0),
431 10, AMR_SUBFRAME_SIZE);
432
433 9136 memcpy(p->pitch_vector, p->excitation, AMR_SUBFRAME_SIZE * sizeof(float));
434 9136 }
435
436 /// @}
437
438
439 /// @name AMR algebraic code book (fixed) vector decoding functions
440 /// @{
441
442 /**
443 * Decode a 10-bit algebraic codebook index from a 10.2 kbit/s frame.
444 */
445 2296 static void decode_10bit_pulse(int code, int pulse_position[8],
446 int i1, int i2, int i3)
447 {
448 // coded using 7+3 bits with the 3 LSBs being, individually, the LSB of 1 of
449 // the 3 pulses and the upper 7 bits being coded in base 5
450 2296 const uint8_t *positions = base_five_table[code >> 3];
451 2296 pulse_position[i1] = (positions[2] << 1) + ( code & 1);
452 2296 pulse_position[i2] = (positions[1] << 1) + ((code >> 1) & 1);
453 2296 pulse_position[i3] = (positions[0] << 1) + ((code >> 2) & 1);
454 2296 }
455
456 /**
457 * Decode the algebraic codebook index to pulse positions and signs and
458 * construct the algebraic codebook vector for MODE_10k2.
459 *
460 * @param fixed_index positions of the eight pulses
461 * @param fixed_sparse pointer to the algebraic codebook vector
462 */
463 1148 static void decode_8_pulses_31bits(const int16_t *fixed_index,
464 AMRFixed *fixed_sparse)
465 {
466 int pulse_position[8];
467 int i, temp;
468
469 1148 decode_10bit_pulse(fixed_index[4], pulse_position, 0, 4, 1);
470 1148 decode_10bit_pulse(fixed_index[5], pulse_position, 2, 6, 5);
471
472 // coded using 5+2 bits with the 2 LSBs being, individually, the LSB of 1 of
473 // the 2 pulses and the upper 5 bits being coded in base 5
474 1148 temp = ((fixed_index[6] >> 2) * 25 + 12) >> 5;
475 1148 pulse_position[3] = temp % 5;
476 1148 pulse_position[7] = temp / 5;
477
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1148 if (pulse_position[7] & 1)
478 480 pulse_position[3] = 4 - pulse_position[3];
479 1148 pulse_position[3] = (pulse_position[3] << 1) + ( fixed_index[6] & 1);
480 1148 pulse_position[7] = (pulse_position[7] << 1) + ((fixed_index[6] >> 1) & 1);
481
482 1148 fixed_sparse->n = 8;
483
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5740 for (i = 0; i < 4; i++) {
484 4592 const int pos1 = (pulse_position[i] << 2) + i;
485 4592 const int pos2 = (pulse_position[i + 4] << 2) + i;
486
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4592 const float sign = fixed_index[i] ? -1.0 : 1.0;
487 4592 fixed_sparse->x[i ] = pos1;
488 4592 fixed_sparse->x[i + 4] = pos2;
489 4592 fixed_sparse->y[i ] = sign;
490
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4592 fixed_sparse->y[i + 4] = pos2 < pos1 ? -sign : sign;
491 }
492 1148 }
493
494 /**
495 * Decode the algebraic codebook index to pulse positions and signs,
496 * then construct the algebraic codebook vector.
497 *
498 * nb of pulses | bits encoding pulses
499 * For MODE_4k75 or MODE_5k15, 2 | 1-3, 4-6, 7
500 * MODE_5k9, 2 | 1, 2-4, 5-6, 7-9
501 * MODE_6k7, 3 | 1-3, 4, 5-7, 8, 9-11
502 * MODE_7k4 or MODE_7k95, 4 | 1-3, 4-6, 7-9, 10, 11-13
503 *
504 * @param fixed_sparse pointer to the algebraic codebook vector
505 * @param pulses algebraic codebook indexes
506 * @param mode mode of the current frame
507 * @param subframe current subframe number
508 */
509 9136 static void decode_fixed_sparse(AMRFixed *fixed_sparse, const uint16_t *pulses,
510 const enum Mode mode, const int subframe)
511 {
512 av_assert1(MODE_4k75 <= (signed)mode && mode <= MODE_12k2);
513
514
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9136 if (mode == MODE_12k2) {
515 1140 ff_decode_10_pulses_35bits(pulses, fixed_sparse, gray_decode, 5, 3);
516
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7996 } else if (mode == MODE_10k2) {
517 1148 decode_8_pulses_31bits(pulses, fixed_sparse);
518 } else {
519 6848 int *pulse_position = fixed_sparse->x;
520 int i, pulse_subset;
521 6848 const int fixed_index = pulses[0];
522
523
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6848 if (mode <= MODE_5k15) {
524 2288 pulse_subset = ((fixed_index >> 3) & 8) + (subframe << 1);
525 2288 pulse_position[0] = ( fixed_index & 7) * 5 + track_position[pulse_subset];
526 2288 pulse_position[1] = ((fixed_index >> 3) & 7) * 5 + track_position[pulse_subset + 1];
527 2288 fixed_sparse->n = 2;
528
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4560 } else if (mode == MODE_5k9) {
529 1140 pulse_subset = ((fixed_index & 1) << 1) + 1;
530 1140 pulse_position[0] = ((fixed_index >> 1) & 7) * 5 + pulse_subset;
531 1140 pulse_subset = (fixed_index >> 4) & 3;
532 1140 pulse_position[1] = ((fixed_index >> 6) & 7) * 5 + pulse_subset + (pulse_subset == 3 ? 1 : 0);
533
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1140 fixed_sparse->n = pulse_position[0] == pulse_position[1] ? 1 : 2;
534
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3420 } else if (mode == MODE_6k7) {
535 1140 pulse_position[0] = (fixed_index & 7) * 5;
536 1140 pulse_subset = (fixed_index >> 2) & 2;
537 1140 pulse_position[1] = ((fixed_index >> 4) & 7) * 5 + pulse_subset + 1;
538 1140 pulse_subset = (fixed_index >> 6) & 2;
539 1140 pulse_position[2] = ((fixed_index >> 8) & 7) * 5 + pulse_subset + 2;
540 1140 fixed_sparse->n = 3;
541 } else { // mode <= MODE_7k95
542 2280 pulse_position[0] = gray_decode[ fixed_index & 7];
543 2280 pulse_position[1] = gray_decode[(fixed_index >> 3) & 7] + 1;
544 2280 pulse_position[2] = gray_decode[(fixed_index >> 6) & 7] + 2;
545 2280 pulse_subset = (fixed_index >> 9) & 1;
546 2280 pulse_position[3] = gray_decode[(fixed_index >> 10) & 7] + pulse_subset + 3;
547 2280 fixed_sparse->n = 4;
548 }
549
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26243 for (i = 0; i < fixed_sparse->n; i++)
550
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19395 fixed_sparse->y[i] = (pulses[1] >> i) & 1 ? 1.0 : -1.0;
551 }
552 9136 }
553
554 /**
555 * Apply pitch lag to obtain the sharpened fixed vector (section 6.1.2)
556 *
557 * @param p the context
558 * @param subframe unpacked amr subframe
559 * @param mode mode of the current frame
560 * @param fixed_sparse sparse representation of the fixed vector
561 */
562 9136 static void pitch_sharpening(AMRContext *p, int subframe, enum Mode mode,
563 AMRFixed *fixed_sparse)
564 {
565 // The spec suggests the current pitch gain is always used, but in other
566 // modes the pitch and codebook gains are jointly quantized (sec 5.8.2)
567 // so the codebook gain cannot depend on the quantized pitch gain.
568
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9136 if (mode == MODE_12k2)
569
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1140 p->beta = FFMIN(p->pitch_gain[4], 1.0);
570
571 9136 fixed_sparse->pitch_lag = p->pitch_lag_int;
572 9136 fixed_sparse->pitch_fac = p->beta;
573
574 // Save pitch sharpening factor for the next subframe
575 // MODE_4k75 only updates on the 2nd and 4th subframes - this follows from
576 // the fact that the gains for two subframes are jointly quantized.
577
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9136 if (mode != MODE_4k75 || subframe & 1)
578 8562 p->beta = av_clipf(p->pitch_gain[4], 0.0, SHARP_MAX);
579 9136 }
580 /// @}
581
582
583 /// @name AMR gain decoding functions
584 /// @{
585
586 /**
587 * fixed gain smoothing
588 * Note that where the spec specifies the "spectrum in the q domain"
589 * in section 6.1.4, in fact frequencies should be used.
590 *
591 * @param p the context
592 * @param lsf LSFs for the current subframe, in the range [0,1]
593 * @param lsf_avg averaged LSFs
594 * @param mode mode of the current frame
595 *
596 * @return fixed gain smoothed
597 */
598 9136 static float fixed_gain_smooth(AMRContext *p , const float *lsf,
599 const float *lsf_avg, const enum Mode mode)
600 {
601 9136 float diff = 0.0;
602 int i;
603
604
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100496 for (i = 0; i < LP_FILTER_ORDER; i++)
605 91360 diff += fabs(lsf_avg[i] - lsf[i]) / lsf_avg[i];
606
607 // If diff is large for ten subframes, disable smoothing for a 40-subframe
608 // hangover period.
609 9136 p->diff_count++;
610
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9136 if (diff <= 0.65)
611 6733 p->diff_count = 0;
612
613
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9136 if (p->diff_count > 10) {
614 687 p->hang_count = 0;
615 687 p->diff_count--; // don't let diff_count overflow
616 }
617
618
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9136 if (p->hang_count < 40) {
619 3851 p->hang_count++;
620
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5285 } else if (mode < MODE_7k4 || mode == MODE_10k2) {
621 3413 const float smoothing_factor = av_clipf(4.0 * diff - 1.6, 0.0, 1.0);
622 3413 const float fixed_gain_mean = (p->fixed_gain[0] + p->fixed_gain[1] +
623 3413 p->fixed_gain[2] + p->fixed_gain[3] +
624 3413 p->fixed_gain[4]) * 0.2;
625 3413 return smoothing_factor * p->fixed_gain[4] +
626 3413 (1.0 - smoothing_factor) * fixed_gain_mean;
627 }
628 5723 return p->fixed_gain[4];
629 }
630
631 /**
632 * Decode pitch gain and fixed gain factor (part of section 6.1.3).
633 *
634 * @param p the context
635 * @param amr_subframe unpacked amr subframe
636 * @param mode mode of the current frame
637 * @param subframe current subframe number
638 * @param fixed_gain_factor decoded gain correction factor
639 */
640 9136 static void decode_gains(AMRContext *p, const AMRNBSubframe *amr_subframe,
641 const enum Mode mode, const int subframe,
642 float *fixed_gain_factor)
643 {
644
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9136 if (mode == MODE_12k2 || mode == MODE_7k95) {
645 2280 p->pitch_gain[4] = qua_gain_pit [amr_subframe->p_gain ]
646 2280 * (1.0 / 16384.0);
647 2280 *fixed_gain_factor = qua_gain_code[amr_subframe->fixed_gain]
648 2280 * (1.0 / 2048.0);
649 } else {
650 const uint16_t *gains;
651
652
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6856 if (mode >= MODE_6k7) {
653 3428 gains = gains_high[amr_subframe->p_gain];
654
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3428 } else if (mode >= MODE_5k15) {
655 2280 gains = gains_low [amr_subframe->p_gain];
656 } else {
657 // gain index is only coded in subframes 0,2 for MODE_4k75
658 1148 gains = gains_MODE_4k75[(p->frame.subframe[subframe & 2].p_gain << 1) + (subframe & 1)];
659 }
660
661 6856 p->pitch_gain[4] = gains[0] * (1.0 / 16384.0);
662 6856 *fixed_gain_factor = gains[1] * (1.0 / 4096.0);
663 }
664 9136 }
665
666 /// @}
667
668
669 /// @name AMR preprocessing functions
670 /// @{
671
672 /**
673 * Circularly convolve a sparse fixed vector with a phase dispersion impulse
674 * response filter (D.6.2 of G.729 and 6.1.5 of AMR).
675 *
676 * @param out vector with filter applied
677 * @param in source vector
678 * @param filter phase filter coefficients
679 *
680 * out[n] = sum(i,0,len-1){ in[i] * filter[(len + n - i)%len] }
681 */
682 4869 static void apply_ir_filter(float *out, const AMRFixed *in,
683 const float *filter)
684 {
685 float filter1[AMR_SUBFRAME_SIZE], ///< filters at pitch lag*1 and *2
686 filter2[AMR_SUBFRAME_SIZE];
687 4869 int lag = in->pitch_lag;
688 4869 float fac = in->pitch_fac;
689 int i;
690
691
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4869 if (lag < AMR_SUBFRAME_SIZE) {
692 741 ff_celp_circ_addf(filter1, filter, filter, lag, fac,
693 AMR_SUBFRAME_SIZE);
694
695
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741 if (lag < AMR_SUBFRAME_SIZE >> 1)
696 ff_celp_circ_addf(filter2, filter, filter1, lag, fac,
697 AMR_SUBFRAME_SIZE);
698 }
699
700 4869 memset(out, 0, sizeof(float) * AMR_SUBFRAME_SIZE);
701
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17676 for (i = 0; i < in->n; i++) {
702 12807 int x = in->x[i];
703 12807 float y = in->y[i];
704 const float *filterp;
705
706
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12807 if (x >= AMR_SUBFRAME_SIZE - lag) {
707 12481 filterp = filter;
708
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326 } else if (x >= AMR_SUBFRAME_SIZE - (lag << 1)) {
709 326 filterp = filter1;
710 } else
711 filterp = filter2;
712
713 12807 ff_celp_circ_addf(out, out, filterp, x, y, AMR_SUBFRAME_SIZE);
714 }
715 4869 }
716
717 /**
718 * Reduce fixed vector sparseness by smoothing with one of three IR filters.
719 * Also know as "adaptive phase dispersion".
720 *
721 * This implements 3GPP TS 26.090 section 6.1(5).
722 *
723 * @param p the context
724 * @param fixed_sparse algebraic codebook vector
725 * @param fixed_vector unfiltered fixed vector
726 * @param fixed_gain smoothed gain
727 * @param out space for modified vector if necessary
728 */
729 9136 static const float *anti_sparseness(AMRContext *p, AMRFixed *fixed_sparse,
730 const float *fixed_vector,
731 float fixed_gain, float *out)
732 {
733 int ir_filter_nr;
734
735
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9136 if (p->pitch_gain[4] < 0.6) {
736 4165 ir_filter_nr = 0; // strong filtering
737
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4971 } else if (p->pitch_gain[4] < 0.9) {
738 2995 ir_filter_nr = 1; // medium filtering
739 } else
740 1976 ir_filter_nr = 2; // no filtering
741
742 // detect 'onset'
743
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9136 if (fixed_gain > 2.0 * p->prev_sparse_fixed_gain) {
744 232 p->ir_filter_onset = 2;
745
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8904 } else if (p->ir_filter_onset)
746 336 p->ir_filter_onset--;
747
748
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9136 if (!p->ir_filter_onset) {
749 8732 int i, count = 0;
750
751
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52392 for (i = 0; i < 5; i++)
752
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43660 if (p->pitch_gain[i] < 0.6)
753 19971 count++;
754
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8732 if (count > 2)
755 3912 ir_filter_nr = 0;
756
757
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8732 if (ir_filter_nr > p->prev_ir_filter_nr + 1)
758 489 ir_filter_nr--;
759
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404 } else if (ir_filter_nr < 2)
760 262 ir_filter_nr++;
761
762 // Disable filtering for very low level of fixed_gain.
763 // Note this step is not specified in the technical description but is in
764 // the reference source in the function Ph_disp.
765
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9136 if (fixed_gain < 5.0)
766 16 ir_filter_nr = 2;
767
768
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9136 if (p->cur_frame_mode != MODE_7k4 && p->cur_frame_mode < MODE_10k2
769
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5708 && ir_filter_nr < 2) {
770 4869 apply_ir_filter(out, fixed_sparse,
771 4869 (p->cur_frame_mode == MODE_7k95 ?
772
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4869 ir_filters_lookup_MODE_7k95 :
773 4869 ir_filters_lookup)[ir_filter_nr]);
774 4869 fixed_vector = out;
775 }
776
777 // update ir filter strength history
778 9136 p->prev_ir_filter_nr = ir_filter_nr;
779 9136 p->prev_sparse_fixed_gain = fixed_gain;
780
781 9136 return fixed_vector;
782 }
783
784 /// @}
785
786
787 /// @name AMR synthesis functions
788 /// @{
789
790 /**
791 * Conduct 10th order linear predictive coding synthesis.
792 *
793 * @param p pointer to the AMRContext
794 * @param lpc pointer to the LPC coefficients
795 * @param fixed_gain fixed codebook gain for synthesis
796 * @param fixed_vector algebraic codebook vector
797 * @param samples pointer to the output speech samples
798 * @param overflow 16-bit overflow flag
799 */
800 9136 static int synthesis(AMRContext *p, float *lpc,
801 float fixed_gain, const float *fixed_vector,
802 float *samples, uint8_t overflow)
803 {
804 int i;
805 float excitation[AMR_SUBFRAME_SIZE];
806
807 // if an overflow has been detected, the pitch vector is scaled down by a
808 // factor of 4
809
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9136 if (overflow)
810 for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
811 p->pitch_vector[i] *= 0.25;
812
813 9136 p->acelpv_ctx.weighted_vector_sumf(excitation, p->pitch_vector, fixed_vector,
814 p->pitch_gain[4], fixed_gain, AMR_SUBFRAME_SIZE);
815
816 // emphasize pitch vector contribution
817
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9136 if (p->pitch_gain[4] > 0.5 && !overflow) {
818 6236 float energy = p->celpm_ctx.dot_productf(excitation, excitation,
819 AMR_SUBFRAME_SIZE);
820 6236 float pitch_factor =
821 12472 p->pitch_gain[4] *
822 6236 (p->cur_frame_mode == MODE_12k2 ?
823
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11768 0.25 * FFMIN(p->pitch_gain[4], 1.0) :
824
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5532 0.5 * FFMIN(p->pitch_gain[4], SHARP_MAX));
825
826
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255676 for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
827 249440 excitation[i] += pitch_factor * p->pitch_vector[i];
828
829 6236 ff_scale_vector_to_given_sum_of_squares(excitation, excitation, energy,
830 AMR_SUBFRAME_SIZE);
831 }
832
833 9136 p->celpf_ctx.celp_lp_synthesis_filterf(samples, lpc, excitation,
834 AMR_SUBFRAME_SIZE,
835 LP_FILTER_ORDER);
836
837 // detect overflow
838
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374576 for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
839
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365440 if (fabsf(samples[i]) > AMR_SAMPLE_BOUND) {
840 return 1;
841 }
842
843 9136 return 0;
844 }
845
846 /// @}
847
848
849 /// @name AMR update functions
850 /// @{
851
852 /**
853 * Update buffers and history at the end of decoding a subframe.
854 *
855 * @param p pointer to the AMRContext
856 */
857 9136 static void update_state(AMRContext *p)
858 {
859 9136 memcpy(p->prev_lsp_sub4, p->lsp[3], LP_FILTER_ORDER * sizeof(p->lsp[3][0]));
860
861 9136 memmove(&p->excitation_buf[0], &p->excitation_buf[AMR_SUBFRAME_SIZE],
862 (PITCH_DELAY_MAX + LP_FILTER_ORDER + 1) * sizeof(float));
863
864 9136 memmove(&p->pitch_gain[0], &p->pitch_gain[1], 4 * sizeof(float));
865 9136 memmove(&p->fixed_gain[0], &p->fixed_gain[1], 4 * sizeof(float));
866
867 9136 memmove(&p->samples_in[0], &p->samples_in[AMR_SUBFRAME_SIZE],
868 LP_FILTER_ORDER * sizeof(float));
869 9136 }
870
871 /// @}
872
873
874 /// @name AMR Postprocessing functions
875 /// @{
876
877 /**
878 * Get the tilt factor of a formant filter from its transfer function
879 *
880 * @param p The Context
881 * @param lpc_n LP_FILTER_ORDER coefficients of the numerator
882 * @param lpc_d LP_FILTER_ORDER coefficients of the denominator
883 */
884 9136 static float tilt_factor(AMRContext *p, float *lpc_n, float *lpc_d)
885 {
886 float rh0, rh1; // autocorrelation at lag 0 and 1
887
888 // LP_FILTER_ORDER prior zeros are needed for ff_celp_lp_synthesis_filterf
889 9136 float impulse_buffer[LP_FILTER_ORDER + AMR_TILT_RESPONSE] = { 0 };
890 9136 float *hf = impulse_buffer + LP_FILTER_ORDER; // start of impulse response
891
892 9136 hf[0] = 1.0;
893 9136 memcpy(hf + 1, lpc_n, sizeof(float) * LP_FILTER_ORDER);
894 9136 p->celpf_ctx.celp_lp_synthesis_filterf(hf, lpc_d, hf,
895 AMR_TILT_RESPONSE,
896 LP_FILTER_ORDER);
897
898 9136 rh0 = p->celpm_ctx.dot_productf(hf, hf, AMR_TILT_RESPONSE);
899 9136 rh1 = p->celpm_ctx.dot_productf(hf, hf + 1, AMR_TILT_RESPONSE - 1);
900
901 // The spec only specifies this check for 12.2 and 10.2 kbit/s
902 // modes. But in the ref source the tilt is always non-negative.
903
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9136 return rh1 >= 0.0 ? rh1 / rh0 * AMR_TILT_GAMMA_T : 0.0;
904 }
905
906 /**
907 * Perform adaptive post-filtering to enhance the quality of the speech.
908 * See section 6.2.1.
909 *
910 * @param p pointer to the AMRContext
911 * @param lpc interpolated LP coefficients for this subframe
912 * @param buf_out output of the filter
913 */
914 9136 static void postfilter(AMRContext *p, float *lpc, float *buf_out)
915 {
916 int i;
917 9136 float *samples = p->samples_in + LP_FILTER_ORDER; // Start of input
918
919 9136 float speech_gain = p->celpm_ctx.dot_productf(samples, samples,
920 AMR_SUBFRAME_SIZE);
921
922 float pole_out[AMR_SUBFRAME_SIZE + LP_FILTER_ORDER]; // Output of pole filter
923 const float *gamma_n, *gamma_d; // Formant filter factor table
924 float lpc_n[LP_FILTER_ORDER], lpc_d[LP_FILTER_ORDER]; // Transfer function coefficients
925
926
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9136 if (p->cur_frame_mode == MODE_12k2 || p->cur_frame_mode == MODE_10k2) {
927 2288 gamma_n = ff_pow_0_7;
928 2288 gamma_d = ff_pow_0_75;
929 } else {
930 6848 gamma_n = ff_pow_0_55;
931 6848 gamma_d = ff_pow_0_7;
932 }
933
934
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100496 for (i = 0; i < LP_FILTER_ORDER; i++) {
935 91360 lpc_n[i] = lpc[i] * gamma_n[i];
936 91360 lpc_d[i] = lpc[i] * gamma_d[i];
937 }
938
939 9136 memcpy(pole_out, p->postfilter_mem, sizeof(float) * LP_FILTER_ORDER);
940 9136 p->celpf_ctx.celp_lp_synthesis_filterf(pole_out + LP_FILTER_ORDER, lpc_d, samples,
941 AMR_SUBFRAME_SIZE, LP_FILTER_ORDER);
942 9136 memcpy(p->postfilter_mem, pole_out + AMR_SUBFRAME_SIZE,
943 sizeof(float) * LP_FILTER_ORDER);
944
945 9136 p->celpf_ctx.celp_lp_zero_synthesis_filterf(buf_out, lpc_n,
946 pole_out + LP_FILTER_ORDER,
947 AMR_SUBFRAME_SIZE, LP_FILTER_ORDER);
948
949 9136 ff_tilt_compensation(&p->tilt_mem, tilt_factor(p, lpc_n, lpc_d), buf_out,
950 AMR_SUBFRAME_SIZE);
951
952 9136 ff_adaptive_gain_control(buf_out, buf_out, speech_gain, AMR_SUBFRAME_SIZE,
953 AMR_AGC_ALPHA, &p->postfilter_agc);
954 9136 }
955
956 /// @}
957
958 2284 static int amrnb_decode_frame(AVCodecContext *avctx, AVFrame *frame,
959 int *got_frame_ptr, AVPacket *avpkt)
960 {
961
962 2284 AMRChannelsContext *s = avctx->priv_data; // pointer to private data
963 2284 const uint8_t *buf = avpkt->data;
964 2284 int buf_size = avpkt->size;
965 int ret;
966
967 /* get output buffer */
968 2284 frame->nb_samples = AMR_BLOCK_SIZE;
969
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2284 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
970 return ret;
971
972
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4568 for (int ch = 0; ch < avctx->ch_layout.nb_channels; ch++) {
973 2284 AMRContext *p = &s->ch[ch];
974 float fixed_gain_factor;
975 2284 AMRFixed fixed_sparse = {0}; // fixed vector up to anti-sparseness processing
976 float spare_vector[AMR_SUBFRAME_SIZE]; // extra stack space to hold result from anti-sparseness processing
977 float synth_fixed_gain; // the fixed gain that synthesis should use
978 const float *synth_fixed_vector; // pointer to the fixed vector that synthesis should use
979 2284 float *buf_out = (float *)frame->extended_data[ch];
980 int channel_size;
981 int i, subframe;
982
983 2284 p->cur_frame_mode = unpack_bitstream(p, buf, buf_size);
984
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2284 if (p->cur_frame_mode == NO_DATA) {
985 av_log(avctx, AV_LOG_ERROR, "Corrupt bitstream\n");
986 return AVERROR_INVALIDDATA;
987 }
988
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2284 if (p->cur_frame_mode == MODE_DTX) {
989 avpriv_report_missing_feature(avctx, "dtx mode");
990 av_log(avctx, AV_LOG_INFO, "Note: libopencore_amrnb supports dtx\n");
991 return AVERROR_PATCHWELCOME;
992 }
993
994 2284 channel_size = frame_sizes_nb[p->cur_frame_mode] + 1; // +7 for rounding and +8 for TOC
995
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2284 if (p->cur_frame_mode == MODE_12k2) {
996 285 lsf2lsp_5(p);
997 } else
998 1999 lsf2lsp_3(p);
999
1000
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11420 for (i = 0; i < 4; i++)
1001 9136 ff_acelp_lspd2lpc(p->lsp[i], p->lpc[i], 5);
1002
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11420 for (subframe = 0; subframe < 4; subframe++) {
1004 9136 const AMRNBSubframe *amr_subframe = &p->frame.subframe[subframe];
1005
1006 9136 decode_pitch_vector(p, amr_subframe, subframe);
1007
1008 9136 decode_fixed_sparse(&fixed_sparse, amr_subframe->pulses,
1009 p->cur_frame_mode, subframe);
1010
1011 // The fixed gain (section 6.1.3) depends on the fixed vector
1012 // (section 6.1.2), but the fixed vector calculation uses
1013 // pitch sharpening based on the on the pitch gain (section 6.1.3).
1014 // So the correct order is: pitch gain, pitch sharpening, fixed gain.
1015 9136 decode_gains(p, amr_subframe, p->cur_frame_mode, subframe,
1016 &fixed_gain_factor);
1017
1018 9136 pitch_sharpening(p, subframe, p->cur_frame_mode, &fixed_sparse);
1019
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9136 if (fixed_sparse.pitch_lag == 0) {
1021 av_log(avctx, AV_LOG_ERROR, "The file is corrupted, pitch_lag = 0 is not allowed\n");
1022 return AVERROR_INVALIDDATA;
1023 }
1024 9136 ff_set_fixed_vector(p->fixed_vector, &fixed_sparse, 1.0,
1025 AMR_SUBFRAME_SIZE);
1026
1027 9136 p->fixed_gain[4] =
1028 9136 ff_amr_set_fixed_gain(fixed_gain_factor,
1029 9136 p->celpm_ctx.dot_productf(p->fixed_vector,
1030 9136 p->fixed_vector,
1031 AMR_SUBFRAME_SIZE) /
1032 AMR_SUBFRAME_SIZE,
1033 9136 p->prediction_error,
1034 9136 energy_mean[p->cur_frame_mode], energy_pred_fac);
1035
1036 // The excitation feedback is calculated without any processing such
1037 // as fixed gain smoothing. This isn't mentioned in the specification.
1038
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374576 for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
1039 365440 p->excitation[i] *= p->pitch_gain[4];
1040 9136 ff_set_fixed_vector(p->excitation, &fixed_sparse, p->fixed_gain[4],
1041 AMR_SUBFRAME_SIZE);
1042
1043 // In the ref decoder, excitation is stored with no fractional bits.
1044 // This step prevents buzz in silent periods. The ref encoder can
1045 // emit long sequences with pitch factor greater than one. This
1046 // creates unwanted feedback if the excitation vector is nonzero.
1047 // (e.g. test sequence T19_795.COD in 3GPP TS 26.074)
1048
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374576 for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
1049 365440 p->excitation[i] = truncf(p->excitation[i]);
1050
1051 // Smooth fixed gain.
1052 // The specification is ambiguous, but in the reference source, the
1053 // smoothed value is NOT fed back into later fixed gain smoothing.
1054 9136 synth_fixed_gain = fixed_gain_smooth(p, p->lsf_q[subframe],
1055 9136 p->lsf_avg, p->cur_frame_mode);
1056
1057 9136 synth_fixed_vector = anti_sparseness(p, &fixed_sparse, p->fixed_vector,
1058 synth_fixed_gain, spare_vector);
1059
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9136 if (synthesis(p, p->lpc[subframe], synth_fixed_gain,
1061 synth_fixed_vector, &p->samples_in[LP_FILTER_ORDER], 0))
1062 // overflow detected -> rerun synthesis scaling pitch vector down
1063 // by a factor of 4, skipping pitch vector contribution emphasis
1064 // and adaptive gain control
1065 synthesis(p, p->lpc[subframe], synth_fixed_gain,
1066 synth_fixed_vector, &p->samples_in[LP_FILTER_ORDER], 1);
1067
1068 9136 postfilter(p, p->lpc[subframe], buf_out + subframe * AMR_SUBFRAME_SIZE);
1069
1070 // update buffers and history
1071 9136 ff_clear_fixed_vector(p->fixed_vector, &fixed_sparse, AMR_SUBFRAME_SIZE);
1072 9136 update_state(p);
1073 }
1074
1075 2284 p->acelpf_ctx.acelp_apply_order_2_transfer_function(buf_out,
1076 buf_out, highpass_zeros,
1077 highpass_poles,
1078 2284 highpass_gain * AMR_SAMPLE_SCALE,
1079 2284 p->high_pass_mem, AMR_BLOCK_SIZE);
1080
1081 /* Update averaged lsf vector (used for fixed gain smoothing).
1082 *
1083 * Note that lsf_avg should not incorporate the current frame's LSFs
1084 * for fixed_gain_smooth.
1085 * The specification has an incorrect formula: the reference decoder uses
1086 * qbar(n-1) rather than qbar(n) in section 6.1(4) equation 71. */
1087 2284 p->acelpv_ctx.weighted_vector_sumf(p->lsf_avg, p->lsf_avg, p->lsf_q[3],
1088 0.84, 0.16, LP_FILTER_ORDER);
1089 2284 buf += channel_size;
1090 2284 buf_size -= channel_size;
1091 }
1092
1093 2284 *got_frame_ptr = 1;
1094
1095 2284 return avpkt->size;
1096 }
1097
1098
1099 const FFCodec ff_amrnb_decoder = {
1100 .p.name = "amrnb",
1101 .p.long_name = NULL_IF_CONFIG_SMALL("AMR-NB (Adaptive Multi-Rate NarrowBand)"),
1102 .p.type = AVMEDIA_TYPE_AUDIO,
1103 .p.id = AV_CODEC_ID_AMR_NB,
1104 .priv_data_size = sizeof(AMRChannelsContext),
1105 .init = amrnb_decode_init,
1106 FF_CODEC_DECODE_CB(amrnb_decode_frame),
1107 .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF,
1108 .p.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP,
1109 AV_SAMPLE_FMT_NONE },
1110 .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
1111 };
1112