FFmpeg coverage


Directory: ../../../ffmpeg/
File: src/libavcodec/amrnbdec.c
Date: 2021-09-24 20:55:06
Exec Total Coverage
Lines: 363 381 95.3%
Branches: 180 198 90.9%

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 "internal.h"
59
60 #include "amrnbdata.h"
61
62 #define AMR_BLOCK_SIZE 160 ///< samples per frame
63 #define AMR_SAMPLE_BOUND 32768.0 ///< threshold for synthesis overflow
64
65 /**
66 * Scale from constructed speech to [-1,1]
67 *
68 * AMR is designed to produce 16-bit PCM samples (3GPP TS 26.090 4.2) but
69 * upscales by two (section 6.2.2).
70 *
71 * Fundamentally, this scale is determined by energy_mean through
72 * the fixed vector contribution to the excitation vector.
73 */
74 #define AMR_SAMPLE_SCALE (2.0 / 32768.0)
75
76 /** Prediction factor for 12.2kbit/s mode */
77 #define PRED_FAC_MODE_12k2 0.65
78
79 #define LSF_R_FAC (8000.0 / 32768.0) ///< LSF residual tables to Hertz
80 #define MIN_LSF_SPACING (50.0488 / 8000.0) ///< Ensures stability of LPC filter
81 #define PITCH_LAG_MIN_MODE_12k2 18 ///< Lower bound on decoded lag search in 12.2kbit/s mode
82
83 /** Initial energy in dB. Also used for bad frames (unimplemented). */
84 #define MIN_ENERGY -14.0
85
86 /** Maximum sharpening factor
87 *
88 * The specification says 0.8, which should be 13107, but the reference C code
89 * uses 13017 instead. (Amusingly the same applies to SHARP_MAX in g729dec.c.)
90 */
91 #define SHARP_MAX 0.79449462890625
92
93 /** Number of impulse response coefficients used for tilt factor */
94 #define AMR_TILT_RESPONSE 22
95 /** Tilt factor = 1st reflection coefficient * gamma_t */
96 #define AMR_TILT_GAMMA_T 0.8
97 /** Adaptive gain control factor used in post-filter */
98 #define AMR_AGC_ALPHA 0.9
99
100 typedef struct AMRContext {
101 AMRNBFrame frame; ///< decoded AMR parameters (lsf coefficients, codebook indexes, etc)
102 uint8_t bad_frame_indicator; ///< bad frame ? 1 : 0
103 enum Mode cur_frame_mode;
104
105 int16_t prev_lsf_r[LP_FILTER_ORDER]; ///< residual LSF vector from previous subframe
106 double lsp[4][LP_FILTER_ORDER]; ///< lsp vectors from current frame
107 double prev_lsp_sub4[LP_FILTER_ORDER]; ///< lsp vector for the 4th subframe of the previous frame
108
109 float lsf_q[4][LP_FILTER_ORDER]; ///< Interpolated LSF vector for fixed gain smoothing
110 float lsf_avg[LP_FILTER_ORDER]; ///< vector of averaged lsf vector
111
112 float lpc[4][LP_FILTER_ORDER]; ///< lpc coefficient vectors for 4 subframes
113
114 uint8_t pitch_lag_int; ///< integer part of pitch lag from current subframe
115
116 float excitation_buf[PITCH_DELAY_MAX + LP_FILTER_ORDER + 1 + AMR_SUBFRAME_SIZE]; ///< current excitation and all necessary excitation history
117 float *excitation; ///< pointer to the current excitation vector in excitation_buf
118
119 float pitch_vector[AMR_SUBFRAME_SIZE]; ///< adaptive code book (pitch) vector
120 float fixed_vector[AMR_SUBFRAME_SIZE]; ///< algebraic codebook (fixed) vector (must be kept zero between frames)
121
122 float prediction_error[4]; ///< quantified prediction errors {20log10(^gamma_gc)} for previous four subframes
123 float pitch_gain[5]; ///< quantified pitch gains for the current and previous four subframes
124 float fixed_gain[5]; ///< quantified fixed gains for the current and previous four subframes
125
126 float beta; ///< previous pitch_gain, bounded by [0.0,SHARP_MAX]
127 uint8_t diff_count; ///< the number of subframes for which diff has been above 0.65
128 uint8_t hang_count; ///< the number of subframes since a hangover period started
129
130 float prev_sparse_fixed_gain; ///< previous fixed gain; used by anti-sparseness processing to determine "onset"
131 uint8_t prev_ir_filter_nr; ///< previous impulse response filter "impNr": 0 - strong, 1 - medium, 2 - none
132 uint8_t ir_filter_onset; ///< flag for impulse response filter strength
133
134 float postfilter_mem[10]; ///< previous intermediate values in the formant filter
135 float tilt_mem; ///< previous input to tilt compensation filter
136 float postfilter_agc; ///< previous factor used for adaptive gain control
137 float high_pass_mem[2]; ///< previous intermediate values in the high-pass filter
138
139 float samples_in[LP_FILTER_ORDER + AMR_SUBFRAME_SIZE]; ///< floating point samples
140
141 ACELPFContext acelpf_ctx; ///< context for filters for ACELP-based codecs
142 ACELPVContext acelpv_ctx; ///< context for vector operations for ACELP-based codecs
143 CELPFContext celpf_ctx; ///< context for filters for CELP-based codecs
144 CELPMContext celpm_ctx; ///< context for fixed point math operations
145
146 } AMRContext;
147
148 /** Double version of ff_weighted_vector_sumf() */
149 570 static void weighted_vector_sumd(double *out, const double *in_a,
150 const double *in_b, double weight_coeff_a,
151 double weight_coeff_b, int length)
152 {
153 int i;
154
155
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156 5700 out[i] = weight_coeff_a * in_a[i]
157 5700 + weight_coeff_b * in_b[i];
158 570 }
159
160 16 static av_cold int amrnb_decode_init(AVCodecContext *avctx)
161 {
162 16 AMRContext *p = avctx->priv_data;
163 int i;
164
165
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16 if (avctx->channels > 1) {
166 avpriv_report_missing_feature(avctx, "multi-channel AMR");
167 return AVERROR_PATCHWELCOME;
168 }
169
170 16 avctx->channels = 1;
171 16 avctx->channel_layout = AV_CH_LAYOUT_MONO;
172
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16 if (!avctx->sample_rate)
173 avctx->sample_rate = 8000;
174 16 avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
175
176 // p->excitation always points to the same position in p->excitation_buf
177 16 p->excitation = &p->excitation_buf[PITCH_DELAY_MAX + LP_FILTER_ORDER + 1];
178
179
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176 for (i = 0; i < LP_FILTER_ORDER; i++) {
180 160 p->prev_lsp_sub4[i] = lsp_sub4_init[i] * 1000 / (float)(1 << 15);
181 160 p->lsf_avg[i] = p->lsf_q[3][i] = lsp_avg_init[i] / (float)(1 << 15);
182 }
183
184
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80 for (i = 0; i < 4; i++)
185 64 p->prediction_error[i] = MIN_ENERGY;
186
187 16 ff_acelp_filter_init(&p->acelpf_ctx);
188 16 ff_acelp_vectors_init(&p->acelpv_ctx);
189 16 ff_celp_filter_init(&p->celpf_ctx);
190 16 ff_celp_math_init(&p->celpm_ctx);
191
192 16 return 0;
193 }
194
195
196 /**
197 * Unpack an RFC4867 speech frame into the AMR frame mode and parameters.
198 *
199 * The order of speech bits is specified by 3GPP TS 26.101.
200 *
201 * @param p the context
202 * @param buf pointer to the input buffer
203 * @param buf_size size of the input buffer
204 *
205 * @return the frame mode
206 */
207 2280 static enum Mode unpack_bitstream(AMRContext *p, const uint8_t *buf,
208 int buf_size)
209 {
210 enum Mode mode;
211
212 // Decode the first octet.
213 2280 mode = buf[0] >> 3 & 0x0F; // frame type
214 2280 p->bad_frame_indicator = (buf[0] & 0x4) != 0x4; // quality bit
215
216
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2280 if (mode >= N_MODES || buf_size < frame_sizes_nb[mode] + 1) {
217 return NO_DATA;
218 }
219
220
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2280 if (mode < MODE_DTX)
221 2280 ff_amr_bit_reorder((uint16_t *) &p->frame, sizeof(AMRNBFrame), buf + 1,
222 2280 amr_unpacking_bitmaps_per_mode[mode]);
223
224 2280 return mode;
225 }
226
227
228 /// @name AMR pitch LPC coefficient decoding functions
229 /// @{
230
231 /**
232 * Interpolate the LSF vector (used for fixed gain smoothing).
233 * The interpolation is done over all four subframes even in MODE_12k2.
234 *
235 * @param[in] ctx The Context
236 * @param[in,out] lsf_q LSFs in [0,1] for each subframe
237 * @param[in] lsf_new New LSFs in [0,1] for subframe 4
238 */
239 2280 static void interpolate_lsf(ACELPVContext *ctx, float lsf_q[4][LP_FILTER_ORDER], float *lsf_new)
240 {
241 int i;
242
243
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11400 for (i = 0; i < 4; i++)
244 9120 ctx->weighted_vector_sumf(lsf_q[i], lsf_q[3], lsf_new,
245 9120 0.25 * (3 - i), 0.25 * (i + 1),
246 LP_FILTER_ORDER);
247 2280 }
248
249 /**
250 * Decode a set of 5 split-matrix quantized lsf indexes into an lsp vector.
251 *
252 * @param p the context
253 * @param lsp output LSP vector
254 * @param lsf_no_r LSF vector without the residual vector added
255 * @param lsf_quantizer pointers to LSF dictionary tables
256 * @param quantizer_offset offset in tables
257 * @param sign for the 3 dictionary table
258 * @param update store data for computing the next frame's LSFs
259 */
260 570 static void lsf2lsp_for_mode12k2(AMRContext *p, double lsp[LP_FILTER_ORDER],
261 const float lsf_no_r[LP_FILTER_ORDER],
262 const int16_t *lsf_quantizer[5],
263 const int quantizer_offset,
264 const int sign, const int update)
265 {
266 int16_t lsf_r[LP_FILTER_ORDER]; // residual LSF vector
267 float lsf_q[LP_FILTER_ORDER]; // quantified LSF vector
268 int i;
269
270
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3420 for (i = 0; i < LP_FILTER_ORDER >> 1; i++)
271 2850 memcpy(&lsf_r[i << 1], &lsf_quantizer[i][quantizer_offset],
272 2 * sizeof(*lsf_r));
273
274
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570 if (sign) {
275 266 lsf_r[4] *= -1;
276 266 lsf_r[5] *= -1;
277 }
278
279
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570 if (update)
280 285 memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(*lsf_r));
281
282
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6270 for (i = 0; i < LP_FILTER_ORDER; i++)
283 5700 lsf_q[i] = lsf_r[i] * (LSF_R_FAC / 8000.0) + lsf_no_r[i] * (1.0 / 8000.0);
284
285 570 ff_set_min_dist_lsf(lsf_q, MIN_LSF_SPACING, LP_FILTER_ORDER);
286
287
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570 if (update)
288 285 interpolate_lsf(&p->acelpv_ctx, p->lsf_q, lsf_q);
289
290 570 ff_acelp_lsf2lspd(lsp, lsf_q, LP_FILTER_ORDER);
291 570 }
292
293 /**
294 * Decode a set of 5 split-matrix quantized lsf indexes into 2 lsp vectors.
295 *
296 * @param p pointer to the AMRContext
297 */
298 285 static void lsf2lsp_5(AMRContext *p)
299 {
300 285 const uint16_t *lsf_param = p->frame.lsf;
301 float lsf_no_r[LP_FILTER_ORDER]; // LSFs without the residual vector
302 const int16_t *lsf_quantizer[5];
303 int i;
304
305 285 lsf_quantizer[0] = lsf_5_1[lsf_param[0]];
306 285 lsf_quantizer[1] = lsf_5_2[lsf_param[1]];
307 285 lsf_quantizer[2] = lsf_5_3[lsf_param[2] >> 1];
308 285 lsf_quantizer[3] = lsf_5_4[lsf_param[3]];
309 285 lsf_quantizer[4] = lsf_5_5[lsf_param[4]];
310
311
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3135 for (i = 0; i < LP_FILTER_ORDER; i++)
312 2850 lsf_no_r[i] = p->prev_lsf_r[i] * LSF_R_FAC * PRED_FAC_MODE_12k2 + lsf_5_mean[i];
313
314 285 lsf2lsp_for_mode12k2(p, p->lsp[1], lsf_no_r, lsf_quantizer, 0, lsf_param[2] & 1, 0);
315 285 lsf2lsp_for_mode12k2(p, p->lsp[3], lsf_no_r, lsf_quantizer, 2, lsf_param[2] & 1, 1);
316
317 // interpolate LSP vectors at subframes 1 and 3
318 285 weighted_vector_sumd(p->lsp[0], p->prev_lsp_sub4, p->lsp[1], 0.5, 0.5, LP_FILTER_ORDER);
319 285 weighted_vector_sumd(p->lsp[2], p->lsp[1] , p->lsp[3], 0.5, 0.5, LP_FILTER_ORDER);
320 285 }
321
322 /**
323 * Decode a set of 3 split-matrix quantized lsf indexes into an lsp vector.
324 *
325 * @param p pointer to the AMRContext
326 */
327 1995 static void lsf2lsp_3(AMRContext *p)
328 {
329 1995 const uint16_t *lsf_param = p->frame.lsf;
330 int16_t lsf_r[LP_FILTER_ORDER]; // residual LSF vector
331 float lsf_q[LP_FILTER_ORDER]; // quantified LSF vector
332 const int16_t *lsf_quantizer;
333 int i, j;
334
335
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1995 lsf_quantizer = (p->cur_frame_mode == MODE_7k95 ? lsf_3_1_MODE_7k95 : lsf_3_1)[lsf_param[0]];
336 1995 memcpy(lsf_r, lsf_quantizer, 3 * sizeof(*lsf_r));
337
338 1995 lsf_quantizer = lsf_3_2[lsf_param[1] << (p->cur_frame_mode <= MODE_5k15)];
339 1995 memcpy(lsf_r + 3, lsf_quantizer, 3 * sizeof(*lsf_r));
340
341
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1995 lsf_quantizer = (p->cur_frame_mode <= MODE_5k15 ? lsf_3_3_MODE_5k15 : lsf_3_3)[lsf_param[2]];
342 1995 memcpy(lsf_r + 6, lsf_quantizer, 4 * sizeof(*lsf_r));
343
344 // calculate mean-removed LSF vector and add mean
345
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21945 for (i = 0; i < LP_FILTER_ORDER; i++)
346 19950 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);
347
348 1995 ff_set_min_dist_lsf(lsf_q, MIN_LSF_SPACING, LP_FILTER_ORDER);
349
350 // store data for computing the next frame's LSFs
351 1995 interpolate_lsf(&p->acelpv_ctx, p->lsf_q, lsf_q);
352 1995 memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(*lsf_r));
353
354 1995 ff_acelp_lsf2lspd(p->lsp[3], lsf_q, LP_FILTER_ORDER);
355
356 // interpolate LSP vectors at subframes 1, 2 and 3
357
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7980 for (i = 1; i <= 3; i++)
358
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65835 for(j = 0; j < LP_FILTER_ORDER; j++)
359 59850 p->lsp[i-1][j] = p->prev_lsp_sub4[j] +
360 59850 (p->lsp[3][j] - p->prev_lsp_sub4[j]) * 0.25 * i;
361 1995 }
362
363 /// @}
364
365
366 /// @name AMR pitch vector decoding functions
367 /// @{
368
369 /**
370 * Like ff_decode_pitch_lag(), but with 1/6 resolution
371 */
372 1140 static void decode_pitch_lag_1_6(int *lag_int, int *lag_frac, int pitch_index,
373 const int prev_lag_int, const int subframe)
374 {
375
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1140 if (subframe == 0 || subframe == 2) {
376
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570 if (pitch_index < 463) {
377 459 *lag_int = (pitch_index + 107) * 10923 >> 16;
378 459 *lag_frac = pitch_index - *lag_int * 6 + 105;
379 } else {
380 111 *lag_int = pitch_index - 368;
381 111 *lag_frac = 0;
382 }
383 } else {
384 570 *lag_int = ((pitch_index + 5) * 10923 >> 16) - 1;
385 570 *lag_frac = pitch_index - *lag_int * 6 - 3;
386 570 *lag_int += av_clip(prev_lag_int - 5, PITCH_LAG_MIN_MODE_12k2,
387 PITCH_DELAY_MAX - 9);
388 }
389 1140 }
390
391 9120 static void decode_pitch_vector(AMRContext *p,
392 const AMRNBSubframe *amr_subframe,
393 const int subframe)
394 {
395 int pitch_lag_int, pitch_lag_frac;
396 9120 enum Mode mode = p->cur_frame_mode;
397
398
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9120 if (p->cur_frame_mode == MODE_12k2) {
399 1140 decode_pitch_lag_1_6(&pitch_lag_int, &pitch_lag_frac,
400 1140 amr_subframe->p_lag, p->pitch_lag_int,
401 subframe);
402 } else {
403
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11400 ff_decode_pitch_lag(&pitch_lag_int, &pitch_lag_frac,
404 7980 amr_subframe->p_lag,
405 7980 p->pitch_lag_int, subframe,
406 mode != MODE_4k75 && mode != MODE_5k15,
407
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3420 mode <= MODE_6k7 ? 4 : (mode == MODE_7k95 ? 5 : 6));
408 7980 pitch_lag_frac *= 2;
409 }
410
411 9120 p->pitch_lag_int = pitch_lag_int; // store previous lag in a uint8_t
412
413 9120 pitch_lag_int += pitch_lag_frac > 0;
414
415 /* Calculate the pitch vector by interpolating the past excitation at the
416 pitch lag using a b60 hamming windowed sinc function. */
417 18240 p->acelpf_ctx.acelp_interpolatef(p->excitation,
418 9120 p->excitation + 1 - pitch_lag_int,
419 ff_b60_sinc, 6,
420
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9120 pitch_lag_frac + 6 - 6*(pitch_lag_frac > 0),
421 10, AMR_SUBFRAME_SIZE);
422
423 9120 memcpy(p->pitch_vector, p->excitation, AMR_SUBFRAME_SIZE * sizeof(float));
424 9120 }
425
426 /// @}
427
428
429 /// @name AMR algebraic code book (fixed) vector decoding functions
430 /// @{
431
432 /**
433 * Decode a 10-bit algebraic codebook index from a 10.2 kbit/s frame.
434 */
435 2280 static void decode_10bit_pulse(int code, int pulse_position[8],
436 int i1, int i2, int i3)
437 {
438 // coded using 7+3 bits with the 3 LSBs being, individually, the LSB of 1 of
439 // the 3 pulses and the upper 7 bits being coded in base 5
440 2280 const uint8_t *positions = base_five_table[code >> 3];
441 2280 pulse_position[i1] = (positions[2] << 1) + ( code & 1);
442 2280 pulse_position[i2] = (positions[1] << 1) + ((code >> 1) & 1);
443 2280 pulse_position[i3] = (positions[0] << 1) + ((code >> 2) & 1);
444 2280 }
445
446 /**
447 * Decode the algebraic codebook index to pulse positions and signs and
448 * construct the algebraic codebook vector for MODE_10k2.
449 *
450 * @param fixed_index positions of the eight pulses
451 * @param fixed_sparse pointer to the algebraic codebook vector
452 */
453 1140 static void decode_8_pulses_31bits(const int16_t *fixed_index,
454 AMRFixed *fixed_sparse)
455 {
456 int pulse_position[8];
457 int i, temp;
458
459 1140 decode_10bit_pulse(fixed_index[4], pulse_position, 0, 4, 1);
460 1140 decode_10bit_pulse(fixed_index[5], pulse_position, 2, 6, 5);
461
462 // coded using 5+2 bits with the 2 LSBs being, individually, the LSB of 1 of
463 // the 2 pulses and the upper 5 bits being coded in base 5
464 1140 temp = ((fixed_index[6] >> 2) * 25 + 12) >> 5;
465 1140 pulse_position[3] = temp % 5;
466 1140 pulse_position[7] = temp / 5;
467
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1140 if (pulse_position[7] & 1)
468 478 pulse_position[3] = 4 - pulse_position[3];
469 1140 pulse_position[3] = (pulse_position[3] << 1) + ( fixed_index[6] & 1);
470 1140 pulse_position[7] = (pulse_position[7] << 1) + ((fixed_index[6] >> 1) & 1);
471
472 1140 fixed_sparse->n = 8;
473
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5700 for (i = 0; i < 4; i++) {
474 4560 const int pos1 = (pulse_position[i] << 2) + i;
475 4560 const int pos2 = (pulse_position[i + 4] << 2) + i;
476
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4560 const float sign = fixed_index[i] ? -1.0 : 1.0;
477 4560 fixed_sparse->x[i ] = pos1;
478 4560 fixed_sparse->x[i + 4] = pos2;
479 4560 fixed_sparse->y[i ] = sign;
480
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4560 fixed_sparse->y[i + 4] = pos2 < pos1 ? -sign : sign;
481 }
482 1140 }
483
484 /**
485 * Decode the algebraic codebook index to pulse positions and signs,
486 * then construct the algebraic codebook vector.
487 *
488 * nb of pulses | bits encoding pulses
489 * For MODE_4k75 or MODE_5k15, 2 | 1-3, 4-6, 7
490 * MODE_5k9, 2 | 1, 2-4, 5-6, 7-9
491 * MODE_6k7, 3 | 1-3, 4, 5-7, 8, 9-11
492 * MODE_7k4 or MODE_7k95, 4 | 1-3, 4-6, 7-9, 10, 11-13
493 *
494 * @param fixed_sparse pointer to the algebraic codebook vector
495 * @param pulses algebraic codebook indexes
496 * @param mode mode of the current frame
497 * @param subframe current subframe number
498 */
499 9120 static void decode_fixed_sparse(AMRFixed *fixed_sparse, const uint16_t *pulses,
500 const enum Mode mode, const int subframe)
501 {
502 av_assert1(MODE_4k75 <= (signed)mode && mode <= MODE_12k2);
503
504
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9120 if (mode == MODE_12k2) {
505 1140 ff_decode_10_pulses_35bits(pulses, fixed_sparse, gray_decode, 5, 3);
506
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7980 } else if (mode == MODE_10k2) {
507 1140 decode_8_pulses_31bits(pulses, fixed_sparse);
508 } else {
509 6840 int *pulse_position = fixed_sparse->x;
510 int i, pulse_subset;
511 6840 const int fixed_index = pulses[0];
512
513
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6840 if (mode <= MODE_5k15) {
514 2280 pulse_subset = ((fixed_index >> 3) & 8) + (subframe << 1);
515 2280 pulse_position[0] = ( fixed_index & 7) * 5 + track_position[pulse_subset];
516 2280 pulse_position[1] = ((fixed_index >> 3) & 7) * 5 + track_position[pulse_subset + 1];
517 2280 fixed_sparse->n = 2;
518
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4560 } else if (mode == MODE_5k9) {
519 1140 pulse_subset = ((fixed_index & 1) << 1) + 1;
520 1140 pulse_position[0] = ((fixed_index >> 1) & 7) * 5 + pulse_subset;
521 1140 pulse_subset = (fixed_index >> 4) & 3;
522 1140 pulse_position[1] = ((fixed_index >> 6) & 7) * 5 + pulse_subset + (pulse_subset == 3 ? 1 : 0);
523
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1140 fixed_sparse->n = pulse_position[0] == pulse_position[1] ? 1 : 2;
524
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3420 } else if (mode == MODE_6k7) {
525 1140 pulse_position[0] = (fixed_index & 7) * 5;
526 1140 pulse_subset = (fixed_index >> 2) & 2;
527 1140 pulse_position[1] = ((fixed_index >> 4) & 7) * 5 + pulse_subset + 1;
528 1140 pulse_subset = (fixed_index >> 6) & 2;
529 1140 pulse_position[2] = ((fixed_index >> 8) & 7) * 5 + pulse_subset + 2;
530 1140 fixed_sparse->n = 3;
531 } else { // mode <= MODE_7k95
532 2280 pulse_position[0] = gray_decode[ fixed_index & 7];
533 2280 pulse_position[1] = gray_decode[(fixed_index >> 3) & 7] + 1;
534 2280 pulse_position[2] = gray_decode[(fixed_index >> 6) & 7] + 2;
535 2280 pulse_subset = (fixed_index >> 9) & 1;
536 2280 pulse_position[3] = gray_decode[(fixed_index >> 10) & 7] + pulse_subset + 3;
537 2280 fixed_sparse->n = 4;
538 }
539
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26219 for (i = 0; i < fixed_sparse->n; i++)
540
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19379 fixed_sparse->y[i] = (pulses[1] >> i) & 1 ? 1.0 : -1.0;
541 }
542 9120 }
543
544 /**
545 * Apply pitch lag to obtain the sharpened fixed vector (section 6.1.2)
546 *
547 * @param p the context
548 * @param subframe unpacked amr subframe
549 * @param mode mode of the current frame
550 * @param fixed_sparse sparse representation of the fixed vector
551 */
552 9120 static void pitch_sharpening(AMRContext *p, int subframe, enum Mode mode,
553 AMRFixed *fixed_sparse)
554 {
555 // The spec suggests the current pitch gain is always used, but in other
556 // modes the pitch and codebook gains are jointly quantized (sec 5.8.2)
557 // so the codebook gain cannot depend on the quantized pitch gain.
558
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9120 if (mode == MODE_12k2)
559
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1140 p->beta = FFMIN(p->pitch_gain[4], 1.0);
560
561 9120 fixed_sparse->pitch_lag = p->pitch_lag_int;
562 9120 fixed_sparse->pitch_fac = p->beta;
563
564 // Save pitch sharpening factor for the next subframe
565 // MODE_4k75 only updates on the 2nd and 4th subframes - this follows from
566 // the fact that the gains for two subframes are jointly quantized.
567
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9120 if (mode != MODE_4k75 || subframe & 1)
568 8550 p->beta = av_clipf(p->pitch_gain[4], 0.0, SHARP_MAX);
569 9120 }
570 /// @}
571
572
573 /// @name AMR gain decoding functions
574 /// @{
575
576 /**
577 * fixed gain smoothing
578 * Note that where the spec specifies the "spectrum in the q domain"
579 * in section 6.1.4, in fact frequencies should be used.
580 *
581 * @param p the context
582 * @param lsf LSFs for the current subframe, in the range [0,1]
583 * @param lsf_avg averaged LSFs
584 * @param mode mode of the current frame
585 *
586 * @return fixed gain smoothed
587 */
588 9120 static float fixed_gain_smooth(AMRContext *p , const float *lsf,
589 const float *lsf_avg, const enum Mode mode)
590 {
591 9120 float diff = 0.0;
592 int i;
593
594
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100320 for (i = 0; i < LP_FILTER_ORDER; i++)
595 91200 diff += fabs(lsf_avg[i] - lsf[i]) / lsf_avg[i];
596
597 // If diff is large for ten subframes, disable smoothing for a 40-subframe
598 // hangover period.
599 9120 p->diff_count++;
600
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9120 if (diff <= 0.65)
601 6717 p->diff_count = 0;
602
603
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9120 if (p->diff_count > 10) {
604 687 p->hang_count = 0;
605 687 p->diff_count--; // don't let diff_count overflow
606 }
607
608
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9120 if (p->hang_count < 40) {
609 3835 p->hang_count++;
610
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5285 } else if (mode < MODE_7k4 || mode == MODE_10k2) {
611 3413 const float smoothing_factor = av_clipf(4.0 * diff - 1.6, 0.0, 1.0);
612 3413 const float fixed_gain_mean = (p->fixed_gain[0] + p->fixed_gain[1] +
613 3413 p->fixed_gain[2] + p->fixed_gain[3] +
614 3413 p->fixed_gain[4]) * 0.2;
615 3413 return smoothing_factor * p->fixed_gain[4] +
616 3413 (1.0 - smoothing_factor) * fixed_gain_mean;
617 }
618 5707 return p->fixed_gain[4];
619 }
620
621 /**
622 * Decode pitch gain and fixed gain factor (part of section 6.1.3).
623 *
624 * @param p the context
625 * @param amr_subframe unpacked amr subframe
626 * @param mode mode of the current frame
627 * @param subframe current subframe number
628 * @param fixed_gain_factor decoded gain correction factor
629 */
630 9120 static void decode_gains(AMRContext *p, const AMRNBSubframe *amr_subframe,
631 const enum Mode mode, const int subframe,
632 float *fixed_gain_factor)
633 {
634
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9120 if (mode == MODE_12k2 || mode == MODE_7k95) {
635 2280 p->pitch_gain[4] = qua_gain_pit [amr_subframe->p_gain ]
636 2280 * (1.0 / 16384.0);
637 2280 *fixed_gain_factor = qua_gain_code[amr_subframe->fixed_gain]
638 2280 * (1.0 / 2048.0);
639 } else {
640 const uint16_t *gains;
641
642
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6840 if (mode >= MODE_6k7) {
643 3420 gains = gains_high[amr_subframe->p_gain];
644
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3420 } else if (mode >= MODE_5k15) {
645 2280 gains = gains_low [amr_subframe->p_gain];
646 } else {
647 // gain index is only coded in subframes 0,2 for MODE_4k75
648 1140 gains = gains_MODE_4k75[(p->frame.subframe[subframe & 2].p_gain << 1) + (subframe & 1)];
649 }
650
651 6840 p->pitch_gain[4] = gains[0] * (1.0 / 16384.0);
652 6840 *fixed_gain_factor = gains[1] * (1.0 / 4096.0);
653 }
654 9120 }
655
656 /// @}
657
658
659 /// @name AMR preprocessing functions
660 /// @{
661
662 /**
663 * Circularly convolve a sparse fixed vector with a phase dispersion impulse
664 * response filter (D.6.2 of G.729 and 6.1.5 of AMR).
665 *
666 * @param out vector with filter applied
667 * @param in source vector
668 * @param filter phase filter coefficients
669 *
670 * out[n] = sum(i,0,len-1){ in[i] * filter[(len + n - i)%len] }
671 */
672 4865 static void apply_ir_filter(float *out, const AMRFixed *in,
673 const float *filter)
674 {
675 float filter1[AMR_SUBFRAME_SIZE], ///< filters at pitch lag*1 and *2
676 filter2[AMR_SUBFRAME_SIZE];
677 4865 int lag = in->pitch_lag;
678 4865 float fac = in->pitch_fac;
679 int i;
680
681
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4865 if (lag < AMR_SUBFRAME_SIZE) {
682 737 ff_celp_circ_addf(filter1, filter, filter, lag, fac,
683 AMR_SUBFRAME_SIZE);
684
685
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737 if (lag < AMR_SUBFRAME_SIZE >> 1)
686 ff_celp_circ_addf(filter2, filter, filter1, lag, fac,
687 AMR_SUBFRAME_SIZE);
688 }
689
690 4865 memset(out, 0, sizeof(float) * AMR_SUBFRAME_SIZE);
691
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17664 for (i = 0; i < in->n; i++) {
692 12799 int x = in->x[i];
693 12799 float y = in->y[i];
694 const float *filterp;
695
696
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12799 if (x >= AMR_SUBFRAME_SIZE - lag) {
697 12477 filterp = filter;
698
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322 } else if (x >= AMR_SUBFRAME_SIZE - (lag << 1)) {
699 322 filterp = filter1;
700 } else
701 filterp = filter2;
702
703 12799 ff_celp_circ_addf(out, out, filterp, x, y, AMR_SUBFRAME_SIZE);
704 }
705 4865 }
706
707 /**
708 * Reduce fixed vector sparseness by smoothing with one of three IR filters.
709 * Also know as "adaptive phase dispersion".
710 *
711 * This implements 3GPP TS 26.090 section 6.1(5).
712 *
713 * @param p the context
714 * @param fixed_sparse algebraic codebook vector
715 * @param fixed_vector unfiltered fixed vector
716 * @param fixed_gain smoothed gain
717 * @param out space for modified vector if necessary
718 */
719 9120 static const float *anti_sparseness(AMRContext *p, AMRFixed *fixed_sparse,
720 const float *fixed_vector,
721 float fixed_gain, float *out)
722 {
723 int ir_filter_nr;
724
725
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9120 if (p->pitch_gain[4] < 0.6) {
726 4157 ir_filter_nr = 0; // strong filtering
727
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4963 } else if (p->pitch_gain[4] < 0.9) {
728 2991 ir_filter_nr = 1; // medium filtering
729 } else
730 1972 ir_filter_nr = 2; // no filtering
731
732 // detect 'onset'
733
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9120 if (fixed_gain > 2.0 * p->prev_sparse_fixed_gain) {
734 218 p->ir_filter_onset = 2;
735
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8902 } else if (p->ir_filter_onset)
736 334 p->ir_filter_onset--;
737
738
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9120 if (!p->ir_filter_onset) {
739 8732 int i, count = 0;
740
741
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52392 for (i = 0; i < 5; i++)
742
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43660 if (p->pitch_gain[i] < 0.6)
743 19971 count++;
744
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8732 if (count > 2)
745 3912 ir_filter_nr = 0;
746
747
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8732 if (ir_filter_nr > p->prev_ir_filter_nr + 1)
748 489 ir_filter_nr--;
749
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388 } else if (ir_filter_nr < 2)
750 250 ir_filter_nr++;
751
752 // Disable filtering for very low level of fixed_gain.
753 // Note this step is not specified in the technical description but is in
754 // the reference source in the function Ph_disp.
755
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9120 if (fixed_gain < 5.0)
756 14 ir_filter_nr = 2;
757
758
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9120 if (p->cur_frame_mode != MODE_7k4 && p->cur_frame_mode < MODE_10k2
759
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5700 && ir_filter_nr < 2) {
760 4865 apply_ir_filter(out, fixed_sparse,
761 4865 (p->cur_frame_mode == MODE_7k95 ?
762
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4865 ir_filters_lookup_MODE_7k95 :
763 4865 ir_filters_lookup)[ir_filter_nr]);
764 4865 fixed_vector = out;
765 }
766
767 // update ir filter strength history
768 9120 p->prev_ir_filter_nr = ir_filter_nr;
769 9120 p->prev_sparse_fixed_gain = fixed_gain;
770
771 9120 return fixed_vector;
772 }
773
774 /// @}
775
776
777 /// @name AMR synthesis functions
778 /// @{
779
780 /**
781 * Conduct 10th order linear predictive coding synthesis.
782 *
783 * @param p pointer to the AMRContext
784 * @param lpc pointer to the LPC coefficients
785 * @param fixed_gain fixed codebook gain for synthesis
786 * @param fixed_vector algebraic codebook vector
787 * @param samples pointer to the output speech samples
788 * @param overflow 16-bit overflow flag
789 */
790 9120 static int synthesis(AMRContext *p, float *lpc,
791 float fixed_gain, const float *fixed_vector,
792 float *samples, uint8_t overflow)
793 {
794 int i;
795 float excitation[AMR_SUBFRAME_SIZE];
796
797 // if an overflow has been detected, the pitch vector is scaled down by a
798 // factor of 4
799
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9120 if (overflow)
800 for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
801 p->pitch_vector[i] *= 0.25;
802
803 9120 p->acelpv_ctx.weighted_vector_sumf(excitation, p->pitch_vector, fixed_vector,
804 p->pitch_gain[4], fixed_gain, AMR_SUBFRAME_SIZE);
805
806 // emphasize pitch vector contribution
807
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9120 if (p->pitch_gain[4] > 0.5 && !overflow) {
808 6222 float energy = p->celpm_ctx.dot_productf(excitation, excitation,
809 AMR_SUBFRAME_SIZE);
810 6222 float pitch_factor =
811 12444 p->pitch_gain[4] *
812 6222 (p->cur_frame_mode == MODE_12k2 ?
813
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11740 0.25 * FFMIN(p->pitch_gain[4], 1.0) :
814
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5518 0.5 * FFMIN(p->pitch_gain[4], SHARP_MAX));
815
816
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255102 for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
817 248880 excitation[i] += pitch_factor * p->pitch_vector[i];
818
819 6222 ff_scale_vector_to_given_sum_of_squares(excitation, excitation, energy,
820 AMR_SUBFRAME_SIZE);
821 }
822
823 9120 p->celpf_ctx.celp_lp_synthesis_filterf(samples, lpc, excitation,
824 AMR_SUBFRAME_SIZE,
825 LP_FILTER_ORDER);
826
827 // detect overflow
828
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373920 for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
829
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364800 if (fabsf(samples[i]) > AMR_SAMPLE_BOUND) {
830 return 1;
831 }
832
833 9120 return 0;
834 }
835
836 /// @}
837
838
839 /// @name AMR update functions
840 /// @{
841
842 /**
843 * Update buffers and history at the end of decoding a subframe.
844 *
845 * @param p pointer to the AMRContext
846 */
847 9120 static void update_state(AMRContext *p)
848 {
849 9120 memcpy(p->prev_lsp_sub4, p->lsp[3], LP_FILTER_ORDER * sizeof(p->lsp[3][0]));
850
851 9120 memmove(&p->excitation_buf[0], &p->excitation_buf[AMR_SUBFRAME_SIZE],
852 (PITCH_DELAY_MAX + LP_FILTER_ORDER + 1) * sizeof(float));
853
854 9120 memmove(&p->pitch_gain[0], &p->pitch_gain[1], 4 * sizeof(float));
855 9120 memmove(&p->fixed_gain[0], &p->fixed_gain[1], 4 * sizeof(float));
856
857 9120 memmove(&p->samples_in[0], &p->samples_in[AMR_SUBFRAME_SIZE],
858 LP_FILTER_ORDER * sizeof(float));
859 9120 }
860
861 /// @}
862
863
864 /// @name AMR Postprocessing functions
865 /// @{
866
867 /**
868 * Get the tilt factor of a formant filter from its transfer function
869 *
870 * @param p The Context
871 * @param lpc_n LP_FILTER_ORDER coefficients of the numerator
872 * @param lpc_d LP_FILTER_ORDER coefficients of the denominator
873 */
874 9120 static float tilt_factor(AMRContext *p, float *lpc_n, float *lpc_d)
875 {
876 float rh0, rh1; // autocorrelation at lag 0 and 1
877
878 // LP_FILTER_ORDER prior zeros are needed for ff_celp_lp_synthesis_filterf
879 9120 float impulse_buffer[LP_FILTER_ORDER + AMR_TILT_RESPONSE] = { 0 };
880 9120 float *hf = impulse_buffer + LP_FILTER_ORDER; // start of impulse response
881
882 9120 hf[0] = 1.0;
883 9120 memcpy(hf + 1, lpc_n, sizeof(float) * LP_FILTER_ORDER);
884 9120 p->celpf_ctx.celp_lp_synthesis_filterf(hf, lpc_d, hf,
885 AMR_TILT_RESPONSE,
886 LP_FILTER_ORDER);
887
888 9120 rh0 = p->celpm_ctx.dot_productf(hf, hf, AMR_TILT_RESPONSE);
889 9120 rh1 = p->celpm_ctx.dot_productf(hf, hf + 1, AMR_TILT_RESPONSE - 1);
890
891 // The spec only specifies this check for 12.2 and 10.2 kbit/s
892 // modes. But in the ref source the tilt is always non-negative.
893
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9120 return rh1 >= 0.0 ? rh1 / rh0 * AMR_TILT_GAMMA_T : 0.0;
894 }
895
896 /**
897 * Perform adaptive post-filtering to enhance the quality of the speech.
898 * See section 6.2.1.
899 *
900 * @param p pointer to the AMRContext
901 * @param lpc interpolated LP coefficients for this subframe
902 * @param buf_out output of the filter
903 */
904 9120 static void postfilter(AMRContext *p, float *lpc, float *buf_out)
905 {
906 int i;
907 9120 float *samples = p->samples_in + LP_FILTER_ORDER; // Start of input
908
909 9120 float speech_gain = p->celpm_ctx.dot_productf(samples, samples,
910 AMR_SUBFRAME_SIZE);
911
912 float pole_out[AMR_SUBFRAME_SIZE + LP_FILTER_ORDER]; // Output of pole filter
913 const float *gamma_n, *gamma_d; // Formant filter factor table
914 float lpc_n[LP_FILTER_ORDER], lpc_d[LP_FILTER_ORDER]; // Transfer function coefficients
915
916
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9120 if (p->cur_frame_mode == MODE_12k2 || p->cur_frame_mode == MODE_10k2) {
917 2280 gamma_n = ff_pow_0_7;
918 2280 gamma_d = ff_pow_0_75;
919 } else {
920 6840 gamma_n = ff_pow_0_55;
921 6840 gamma_d = ff_pow_0_7;
922 }
923
924
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100320 for (i = 0; i < LP_FILTER_ORDER; i++) {
925 91200 lpc_n[i] = lpc[i] * gamma_n[i];
926 91200 lpc_d[i] = lpc[i] * gamma_d[i];
927 }
928
929 9120 memcpy(pole_out, p->postfilter_mem, sizeof(float) * LP_FILTER_ORDER);
930 9120 p->celpf_ctx.celp_lp_synthesis_filterf(pole_out + LP_FILTER_ORDER, lpc_d, samples,
931 AMR_SUBFRAME_SIZE, LP_FILTER_ORDER);
932 9120 memcpy(p->postfilter_mem, pole_out + AMR_SUBFRAME_SIZE,
933 sizeof(float) * LP_FILTER_ORDER);
934
935 9120 p->celpf_ctx.celp_lp_zero_synthesis_filterf(buf_out, lpc_n,
936 pole_out + LP_FILTER_ORDER,
937 AMR_SUBFRAME_SIZE, LP_FILTER_ORDER);
938
939 9120 ff_tilt_compensation(&p->tilt_mem, tilt_factor(p, lpc_n, lpc_d), buf_out,
940 AMR_SUBFRAME_SIZE);
941
942 9120 ff_adaptive_gain_control(buf_out, buf_out, speech_gain, AMR_SUBFRAME_SIZE,
943 AMR_AGC_ALPHA, &p->postfilter_agc);
944 9120 }
945
946 /// @}
947
948 2280 static int amrnb_decode_frame(AVCodecContext *avctx, void *data,
949 int *got_frame_ptr, AVPacket *avpkt)
950 {
951
952 2280 AMRContext *p = avctx->priv_data; // pointer to private data
953 2280 AVFrame *frame = data;
954 2280 const uint8_t *buf = avpkt->data;
955 2280 int buf_size = avpkt->size;
956 float *buf_out; // pointer to the output data buffer
957 int i, subframe, ret;
958 float fixed_gain_factor;
959 2280 AMRFixed fixed_sparse = {0}; // fixed vector up to anti-sparseness processing
960 float spare_vector[AMR_SUBFRAME_SIZE]; // extra stack space to hold result from anti-sparseness processing
961 float synth_fixed_gain; // the fixed gain that synthesis should use
962 const float *synth_fixed_vector; // pointer to the fixed vector that synthesis should use
963
964 /* get output buffer */
965 2280 frame->nb_samples = AMR_BLOCK_SIZE;
966
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2280 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
967 return ret;
968 2280 buf_out = (float *)frame->data[0];
969
970 2280 p->cur_frame_mode = unpack_bitstream(p, buf, buf_size);
971
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2280 if (p->cur_frame_mode == NO_DATA) {
972 av_log(avctx, AV_LOG_ERROR, "Corrupt bitstream\n");
973 return AVERROR_INVALIDDATA;
974 }
975
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2280 if (p->cur_frame_mode == MODE_DTX) {
976 avpriv_report_missing_feature(avctx, "dtx mode");
977 av_log(avctx, AV_LOG_INFO, "Note: libopencore_amrnb supports dtx\n");
978 return AVERROR_PATCHWELCOME;
979 }
980
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2280 if (p->cur_frame_mode == MODE_12k2) {
982 285 lsf2lsp_5(p);
983 } else
984 1995 lsf2lsp_3(p);
985
986
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11400 for (i = 0; i < 4; i++)
987 9120 ff_acelp_lspd2lpc(p->lsp[i], p->lpc[i], 5);
988
989
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11400 for (subframe = 0; subframe < 4; subframe++) {
990 9120 const AMRNBSubframe *amr_subframe = &p->frame.subframe[subframe];
991
992 9120 decode_pitch_vector(p, amr_subframe, subframe);
993
994 9120 decode_fixed_sparse(&fixed_sparse, amr_subframe->pulses,
995 p->cur_frame_mode, subframe);
996
997 // The fixed gain (section 6.1.3) depends on the fixed vector
998 // (section 6.1.2), but the fixed vector calculation uses
999 // pitch sharpening based on the on the pitch gain (section 6.1.3).
1000 // So the correct order is: pitch gain, pitch sharpening, fixed gain.
1001 9120 decode_gains(p, amr_subframe, p->cur_frame_mode, subframe,
1002 &fixed_gain_factor);
1003
1004 9120 pitch_sharpening(p, subframe, p->cur_frame_mode, &fixed_sparse);
1005
1006
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9120 if (fixed_sparse.pitch_lag == 0) {
1007 av_log(avctx, AV_LOG_ERROR, "The file is corrupted, pitch_lag = 0 is not allowed\n");
1008 return AVERROR_INVALIDDATA;
1009 }
1010 9120 ff_set_fixed_vector(p->fixed_vector, &fixed_sparse, 1.0,
1011 AMR_SUBFRAME_SIZE);
1012
1013 9120 p->fixed_gain[4] =
1014 9120 ff_amr_set_fixed_gain(fixed_gain_factor,
1015 9120 p->celpm_ctx.dot_productf(p->fixed_vector,
1016 9120 p->fixed_vector,
1017 AMR_SUBFRAME_SIZE) /
1018 AMR_SUBFRAME_SIZE,
1019 9120 p->prediction_error,
1020 9120 energy_mean[p->cur_frame_mode], energy_pred_fac);
1021
1022 // The excitation feedback is calculated without any processing such
1023 // as fixed gain smoothing. This isn't mentioned in the specification.
1024
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373920 for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
1025 364800 p->excitation[i] *= p->pitch_gain[4];
1026 9120 ff_set_fixed_vector(p->excitation, &fixed_sparse, p->fixed_gain[4],
1027 AMR_SUBFRAME_SIZE);
1028
1029 // In the ref decoder, excitation is stored with no fractional bits.
1030 // This step prevents buzz in silent periods. The ref encoder can
1031 // emit long sequences with pitch factor greater than one. This
1032 // creates unwanted feedback if the excitation vector is nonzero.
1033 // (e.g. test sequence T19_795.COD in 3GPP TS 26.074)
1034
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373920 for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
1035 364800 p->excitation[i] = truncf(p->excitation[i]);
1036
1037 // Smooth fixed gain.
1038 // The specification is ambiguous, but in the reference source, the
1039 // smoothed value is NOT fed back into later fixed gain smoothing.
1040 9120 synth_fixed_gain = fixed_gain_smooth(p, p->lsf_q[subframe],
1041 9120 p->lsf_avg, p->cur_frame_mode);
1042
1043 9120 synth_fixed_vector = anti_sparseness(p, &fixed_sparse, p->fixed_vector,
1044 synth_fixed_gain, spare_vector);
1045
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9120 if (synthesis(p, p->lpc[subframe], synth_fixed_gain,
1047 synth_fixed_vector, &p->samples_in[LP_FILTER_ORDER], 0))
1048 // overflow detected -> rerun synthesis scaling pitch vector down
1049 // by a factor of 4, skipping pitch vector contribution emphasis
1050 // and adaptive gain control
1051 synthesis(p, p->lpc[subframe], synth_fixed_gain,
1052 synth_fixed_vector, &p->samples_in[LP_FILTER_ORDER], 1);
1053
1054 9120 postfilter(p, p->lpc[subframe], buf_out + subframe * AMR_SUBFRAME_SIZE);
1055
1056 // update buffers and history
1057 9120 ff_clear_fixed_vector(p->fixed_vector, &fixed_sparse, AMR_SUBFRAME_SIZE);
1058 9120 update_state(p);
1059 }
1060
1061 2280 p->acelpf_ctx.acelp_apply_order_2_transfer_function(buf_out,
1062 buf_out, highpass_zeros,
1063 highpass_poles,
1064 2280 highpass_gain * AMR_SAMPLE_SCALE,
1065 2280 p->high_pass_mem, AMR_BLOCK_SIZE);
1066
1067 /* Update averaged lsf vector (used for fixed gain smoothing).
1068 *
1069 * Note that lsf_avg should not incorporate the current frame's LSFs
1070 * for fixed_gain_smooth.
1071 * The specification has an incorrect formula: the reference decoder uses
1072 * qbar(n-1) rather than qbar(n) in section 6.1(4) equation 71. */
1073 2280 p->acelpv_ctx.weighted_vector_sumf(p->lsf_avg, p->lsf_avg, p->lsf_q[3],
1074 0.84, 0.16, LP_FILTER_ORDER);
1075
1076 2280 *got_frame_ptr = 1;
1077
1078 /* return the amount of bytes consumed if everything was OK */
1079 2280 return frame_sizes_nb[p->cur_frame_mode] + 1; // +7 for rounding and +8 for TOC
1080 }
1081
1082
1083 const AVCodec ff_amrnb_decoder = {
1084 .name = "amrnb",
1085 .long_name = NULL_IF_CONFIG_SMALL("AMR-NB (Adaptive Multi-Rate NarrowBand)"),
1086 .type = AVMEDIA_TYPE_AUDIO,
1087 .id = AV_CODEC_ID_AMR_NB,
1088 .priv_data_size = sizeof(AMRContext),
1089 .init = amrnb_decode_init,
1090 .decode = amrnb_decode_frame,
1091 .capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF,
1092 .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT,
1093 AV_SAMPLE_FMT_NONE },
1094 };
1095