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amrnbdec.c
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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 
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
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 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  for (i = 0; i < length; i++)
156  out[i] = weight_coeff_a * in_a[i]
157  + weight_coeff_b * in_b[i];
158 }
159 
161 {
162  AMRContext *p = avctx->priv_data;
163  int i;
164 
165  if (avctx->channels > 1) {
166  av_log_missing_feature(avctx, "multi-channel AMR", 0);
167  return AVERROR_PATCHWELCOME;
168  }
169 
170  avctx->channels = 1;
172  if (!avctx->sample_rate)
173  avctx->sample_rate = 8000;
174  avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
175 
176  // p->excitation always points to the same position in p->excitation_buf
178 
179  for (i = 0; i < LP_FILTER_ORDER; i++) {
180  p->prev_lsp_sub4[i] = lsp_sub4_init[i] * 1000 / (float)(1 << 15);
181  p->lsf_avg[i] = p->lsf_q[3][i] = lsp_avg_init[i] / (float)(1 << 15);
182  }
183 
184  for (i = 0; i < 4; i++)
186 
191 
192  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 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  mode = buf[0] >> 3 & 0x0F; // frame type
214  p->bad_frame_indicator = (buf[0] & 0x4) != 0x4; // quality bit
215 
216  if (mode >= N_MODES || buf_size < frame_sizes_nb[mode] + 1) {
217  return NO_DATA;
218  }
219 
220  if (mode < MODE_DTX)
221  ff_amr_bit_reorder((uint16_t *) &p->frame, sizeof(AMRNBFrame), buf + 1,
223 
224  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 static void interpolate_lsf(ACELPVContext *ctx, float lsf_q[4][LP_FILTER_ORDER], float *lsf_new)
240 {
241  int i;
242 
243  for (i = 0; i < 4; i++)
244  ctx->weighted_vector_sumf(lsf_q[i], lsf_q[3], lsf_new,
245  0.25 * (3 - i), 0.25 * (i + 1),
246  LP_FILTER_ORDER);
247 }
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 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  for (i = 0; i < LP_FILTER_ORDER >> 1; i++)
271  memcpy(&lsf_r[i << 1], &lsf_quantizer[i][quantizer_offset],
272  2 * sizeof(*lsf_r));
273 
274  if (sign) {
275  lsf_r[4] *= -1;
276  lsf_r[5] *= -1;
277  }
278 
279  if (update)
280  memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(*lsf_r));
281 
282  for (i = 0; i < LP_FILTER_ORDER; i++)
283  lsf_q[i] = lsf_r[i] * (LSF_R_FAC / 8000.0) + lsf_no_r[i] * (1.0 / 8000.0);
284 
285  ff_set_min_dist_lsf(lsf_q, MIN_LSF_SPACING, LP_FILTER_ORDER);
286 
287  if (update)
288  interpolate_lsf(&p->acelpv_ctx, p->lsf_q, lsf_q);
289 
290  ff_acelp_lsf2lspd(lsp, lsf_q, LP_FILTER_ORDER);
291 }
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 static void lsf2lsp_5(AMRContext *p)
299 {
300  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  lsf_quantizer[0] = lsf_5_1[lsf_param[0]];
306  lsf_quantizer[1] = lsf_5_2[lsf_param[1]];
307  lsf_quantizer[2] = lsf_5_3[lsf_param[2] >> 1];
308  lsf_quantizer[3] = lsf_5_4[lsf_param[3]];
309  lsf_quantizer[4] = lsf_5_5[lsf_param[4]];
310 
311  for (i = 0; i < LP_FILTER_ORDER; i++)
312  lsf_no_r[i] = p->prev_lsf_r[i] * LSF_R_FAC * PRED_FAC_MODE_12k2 + lsf_5_mean[i];
313 
314  lsf2lsp_for_mode12k2(p, p->lsp[1], lsf_no_r, lsf_quantizer, 0, lsf_param[2] & 1, 0);
315  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  weighted_vector_sumd(p->lsp[0], p->prev_lsp_sub4, p->lsp[1], 0.5, 0.5, LP_FILTER_ORDER);
319  weighted_vector_sumd(p->lsp[2], p->lsp[1] , p->lsp[3], 0.5, 0.5, LP_FILTER_ORDER);
320 }
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 static void lsf2lsp_3(AMRContext *p)
328 {
329  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  lsf_quantizer = (p->cur_frame_mode == MODE_7k95 ? lsf_3_1_MODE_7k95 : lsf_3_1)[lsf_param[0]];
336  memcpy(lsf_r, lsf_quantizer, 3 * sizeof(*lsf_r));
337 
338  lsf_quantizer = lsf_3_2[lsf_param[1] << (p->cur_frame_mode <= MODE_5k15)];
339  memcpy(lsf_r + 3, lsf_quantizer, 3 * sizeof(*lsf_r));
340 
341  lsf_quantizer = (p->cur_frame_mode <= MODE_5k15 ? lsf_3_3_MODE_5k15 : lsf_3_3)[lsf_param[2]];
342  memcpy(lsf_r + 6, lsf_quantizer, 4 * sizeof(*lsf_r));
343 
344  // calculate mean-removed LSF vector and add mean
345  for (i = 0; i < LP_FILTER_ORDER; i++)
346  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  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  interpolate_lsf(&p->acelpv_ctx, p->lsf_q, lsf_q);
352  memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(*lsf_r));
353 
354  ff_acelp_lsf2lspd(p->lsp[3], lsf_q, LP_FILTER_ORDER);
355 
356  // interpolate LSP vectors at subframes 1, 2 and 3
357  for (i = 1; i <= 3; i++)
358  for(j = 0; j < LP_FILTER_ORDER; j++)
359  p->lsp[i-1][j] = p->prev_lsp_sub4[j] +
360  (p->lsp[3][j] - p->prev_lsp_sub4[j]) * 0.25 * i;
361 }
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 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  if (subframe == 0 || subframe == 2) {
376  if (pitch_index < 463) {
377  *lag_int = (pitch_index + 107) * 10923 >> 16;
378  *lag_frac = pitch_index - *lag_int * 6 + 105;
379  } else {
380  *lag_int = pitch_index - 368;
381  *lag_frac = 0;
382  }
383  } else {
384  *lag_int = ((pitch_index + 5) * 10923 >> 16) - 1;
385  *lag_frac = pitch_index - *lag_int * 6 - 3;
386  *lag_int += av_clip(prev_lag_int - 5, PITCH_LAG_MIN_MODE_12k2,
387  PITCH_DELAY_MAX - 9);
388  }
389 }
390 
392  const AMRNBSubframe *amr_subframe,
393  const int subframe)
394 {
395  int pitch_lag_int, pitch_lag_frac;
396  enum Mode mode = p->cur_frame_mode;
397 
398  if (p->cur_frame_mode == MODE_12k2) {
399  decode_pitch_lag_1_6(&pitch_lag_int, &pitch_lag_frac,
400  amr_subframe->p_lag, p->pitch_lag_int,
401  subframe);
402  } else
403  ff_decode_pitch_lag(&pitch_lag_int, &pitch_lag_frac,
404  amr_subframe->p_lag,
405  p->pitch_lag_int, subframe,
406  mode != MODE_4k75 && mode != MODE_5k15,
407  mode <= MODE_6k7 ? 4 : (mode == MODE_7k95 ? 5 : 6));
408 
409  p->pitch_lag_int = pitch_lag_int; // store previous lag in a uint8_t
410 
411  pitch_lag_frac <<= (p->cur_frame_mode != MODE_12k2);
412 
413  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. */
418  p->excitation + 1 - pitch_lag_int,
419  ff_b60_sinc, 6,
420  pitch_lag_frac + 6 - 6*(pitch_lag_frac > 0),
421  10, AMR_SUBFRAME_SIZE);
422 
423  memcpy(p->pitch_vector, p->excitation, AMR_SUBFRAME_SIZE * sizeof(float));
424 }
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 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  const uint8_t *positions = base_five_table[code >> 3];
441  pulse_position[i1] = (positions[2] << 1) + ( code & 1);
442  pulse_position[i2] = (positions[1] << 1) + ((code >> 1) & 1);
443  pulse_position[i3] = (positions[0] << 1) + ((code >> 2) & 1);
444 }
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 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  decode_10bit_pulse(fixed_index[4], pulse_position, 0, 4, 1);
460  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  temp = ((fixed_index[6] >> 2) * 25 + 12) >> 5;
465  pulse_position[3] = temp % 5;
466  pulse_position[7] = temp / 5;
467  if (pulse_position[7] & 1)
468  pulse_position[3] = 4 - pulse_position[3];
469  pulse_position[3] = (pulse_position[3] << 1) + ( fixed_index[6] & 1);
470  pulse_position[7] = (pulse_position[7] << 1) + ((fixed_index[6] >> 1) & 1);
471 
472  fixed_sparse->n = 8;
473  for (i = 0; i < 4; i++) {
474  const int pos1 = (pulse_position[i] << 2) + i;
475  const int pos2 = (pulse_position[i + 4] << 2) + i;
476  const float sign = fixed_index[i] ? -1.0 : 1.0;
477  fixed_sparse->x[i ] = pos1;
478  fixed_sparse->x[i + 4] = pos2;
479  fixed_sparse->y[i ] = sign;
480  fixed_sparse->y[i + 4] = pos2 < pos1 ? -sign : sign;
481  }
482 }
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 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  if (mode == MODE_12k2) {
505  ff_decode_10_pulses_35bits(pulses, fixed_sparse, gray_decode, 5, 3);
506  } else if (mode == MODE_10k2) {
507  decode_8_pulses_31bits(pulses, fixed_sparse);
508  } else {
509  int *pulse_position = fixed_sparse->x;
510  int i, pulse_subset;
511  const int fixed_index = pulses[0];
512 
513  if (mode <= MODE_5k15) {
514  pulse_subset = ((fixed_index >> 3) & 8) + (subframe << 1);
515  pulse_position[0] = ( fixed_index & 7) * 5 + track_position[pulse_subset];
516  pulse_position[1] = ((fixed_index >> 3) & 7) * 5 + track_position[pulse_subset + 1];
517  fixed_sparse->n = 2;
518  } else if (mode == MODE_5k9) {
519  pulse_subset = ((fixed_index & 1) << 1) + 1;
520  pulse_position[0] = ((fixed_index >> 1) & 7) * 5 + pulse_subset;
521  pulse_subset = (fixed_index >> 4) & 3;
522  pulse_position[1] = ((fixed_index >> 6) & 7) * 5 + pulse_subset + (pulse_subset == 3 ? 1 : 0);
523  fixed_sparse->n = pulse_position[0] == pulse_position[1] ? 1 : 2;
524  } else if (mode == MODE_6k7) {
525  pulse_position[0] = (fixed_index & 7) * 5;
526  pulse_subset = (fixed_index >> 2) & 2;
527  pulse_position[1] = ((fixed_index >> 4) & 7) * 5 + pulse_subset + 1;
528  pulse_subset = (fixed_index >> 6) & 2;
529  pulse_position[2] = ((fixed_index >> 8) & 7) * 5 + pulse_subset + 2;
530  fixed_sparse->n = 3;
531  } else { // mode <= MODE_7k95
532  pulse_position[0] = gray_decode[ fixed_index & 7];
533  pulse_position[1] = gray_decode[(fixed_index >> 3) & 7] + 1;
534  pulse_position[2] = gray_decode[(fixed_index >> 6) & 7] + 2;
535  pulse_subset = (fixed_index >> 9) & 1;
536  pulse_position[3] = gray_decode[(fixed_index >> 10) & 7] + pulse_subset + 3;
537  fixed_sparse->n = 4;
538  }
539  for (i = 0; i < fixed_sparse->n; i++)
540  fixed_sparse->y[i] = (pulses[1] >> i) & 1 ? 1.0 : -1.0;
541  }
542 }
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 respresentation of the fixed vector
551  */
552 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 joinly quantized (sec 5.8.2)
557  // so the codebook gain cannot depend on the quantized pitch gain.
558  if (mode == MODE_12k2)
559  p->beta = FFMIN(p->pitch_gain[4], 1.0);
560 
561  fixed_sparse->pitch_lag = p->pitch_lag_int;
562  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  if (mode != MODE_4k75 || subframe & 1)
568  p->beta = av_clipf(p->pitch_gain[4], 0.0, SHARP_MAX);
569 }
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 static float fixed_gain_smooth(AMRContext *p , const float *lsf,
589  const float *lsf_avg, const enum Mode mode)
590 {
591  float diff = 0.0;
592  int i;
593 
594  for (i = 0; i < LP_FILTER_ORDER; i++)
595  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  p->diff_count++;
600  if (diff <= 0.65)
601  p->diff_count = 0;
602 
603  if (p->diff_count > 10) {
604  p->hang_count = 0;
605  p->diff_count--; // don't let diff_count overflow
606  }
607 
608  if (p->hang_count < 40) {
609  p->hang_count++;
610  } else if (mode < MODE_7k4 || mode == MODE_10k2) {
611  const float smoothing_factor = av_clipf(4.0 * diff - 1.6, 0.0, 1.0);
612  const float fixed_gain_mean = (p->fixed_gain[0] + p->fixed_gain[1] +
613  p->fixed_gain[2] + p->fixed_gain[3] +
614  p->fixed_gain[4]) * 0.2;
615  return smoothing_factor * p->fixed_gain[4] +
616  (1.0 - smoothing_factor) * fixed_gain_mean;
617  }
618  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 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  if (mode == MODE_12k2 || mode == MODE_7k95) {
635  p->pitch_gain[4] = qua_gain_pit [amr_subframe->p_gain ]
636  * (1.0 / 16384.0);
637  *fixed_gain_factor = qua_gain_code[amr_subframe->fixed_gain]
638  * (1.0 / 2048.0);
639  } else {
640  const uint16_t *gains;
641 
642  if (mode >= MODE_6k7) {
643  gains = gains_high[amr_subframe->p_gain];
644  } else if (mode >= MODE_5k15) {
645  gains = gains_low [amr_subframe->p_gain];
646  } else {
647  // gain index is only coded in subframes 0,2 for MODE_4k75
648  gains = gains_MODE_4k75[(p->frame.subframe[subframe & 2].p_gain << 1) + (subframe & 1)];
649  }
650 
651  p->pitch_gain[4] = gains[0] * (1.0 / 16384.0);
652  *fixed_gain_factor = gains[1] * (1.0 / 4096.0);
653  }
654 }
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 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  int lag = in->pitch_lag;
678  float fac = in->pitch_fac;
679  int i;
680 
681  if (lag < AMR_SUBFRAME_SIZE) {
682  ff_celp_circ_addf(filter1, filter, filter, lag, fac,
684 
685  if (lag < AMR_SUBFRAME_SIZE >> 1)
686  ff_celp_circ_addf(filter2, filter, filter1, lag, fac,
688  }
689 
690  memset(out, 0, sizeof(float) * AMR_SUBFRAME_SIZE);
691  for (i = 0; i < in->n; i++) {
692  int x = in->x[i];
693  float y = in->y[i];
694  const float *filterp;
695 
696  if (x >= AMR_SUBFRAME_SIZE - lag) {
697  filterp = filter;
698  } else if (x >= AMR_SUBFRAME_SIZE - (lag << 1)) {
699  filterp = filter1;
700  } else
701  filterp = filter2;
702 
703  ff_celp_circ_addf(out, out, filterp, x, y, AMR_SUBFRAME_SIZE);
704  }
705 }
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 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  if (p->pitch_gain[4] < 0.6) {
726  ir_filter_nr = 0; // strong filtering
727  } else if (p->pitch_gain[4] < 0.9) {
728  ir_filter_nr = 1; // medium filtering
729  } else
730  ir_filter_nr = 2; // no filtering
731 
732  // detect 'onset'
733  if (fixed_gain > 2.0 * p->prev_sparse_fixed_gain) {
734  p->ir_filter_onset = 2;
735  } else if (p->ir_filter_onset)
736  p->ir_filter_onset--;
737 
738  if (!p->ir_filter_onset) {
739  int i, count = 0;
740 
741  for (i = 0; i < 5; i++)
742  if (p->pitch_gain[i] < 0.6)
743  count++;
744  if (count > 2)
745  ir_filter_nr = 0;
746 
747  if (ir_filter_nr > p->prev_ir_filter_nr + 1)
748  ir_filter_nr--;
749  } else if (ir_filter_nr < 2)
750  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  if (fixed_gain < 5.0)
756  ir_filter_nr = 2;
757 
759  && ir_filter_nr < 2) {
760  apply_ir_filter(out, fixed_sparse,
761  (p->cur_frame_mode == MODE_7k95 ?
763  ir_filters_lookup)[ir_filter_nr]);
764  fixed_vector = out;
765  }
766 
767  // update ir filter strength history
768  p->prev_ir_filter_nr = ir_filter_nr;
769  p->prev_sparse_fixed_gain = fixed_gain;
770 
771  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 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  if (overflow)
800  for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
801  p->pitch_vector[i] *= 0.25;
802 
803  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  if (p->pitch_gain[4] > 0.5 && !overflow) {
808  float energy = p->celpm_ctx.dot_productf(excitation, excitation,
809  AMR_SUBFRAME_SIZE);
810  float pitch_factor =
811  p->pitch_gain[4] *
812  (p->cur_frame_mode == MODE_12k2 ?
813  0.25 * FFMIN(p->pitch_gain[4], 1.0) :
814  0.5 * FFMIN(p->pitch_gain[4], SHARP_MAX));
815 
816  for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
817  excitation[i] += pitch_factor * p->pitch_vector[i];
818 
819  ff_scale_vector_to_given_sum_of_squares(excitation, excitation, energy,
820  AMR_SUBFRAME_SIZE);
821  }
822 
823  p->celpf_ctx.celp_lp_synthesis_filterf(samples, lpc, excitation,
824  AMR_SUBFRAME_SIZE,
826 
827  // detect overflow
828  for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
829  if (fabsf(samples[i]) > AMR_SAMPLE_BOUND) {
830  return 1;
831  }
832 
833  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 static void update_state(AMRContext *p)
848 {
849  memcpy(p->prev_lsp_sub4, p->lsp[3], LP_FILTER_ORDER * sizeof(p->lsp[3][0]));
850 
851  memmove(&p->excitation_buf[0], &p->excitation_buf[AMR_SUBFRAME_SIZE],
852  (PITCH_DELAY_MAX + LP_FILTER_ORDER + 1) * sizeof(float));
853 
854  memmove(&p->pitch_gain[0], &p->pitch_gain[1], 4 * sizeof(float));
855  memmove(&p->fixed_gain[0], &p->fixed_gain[1], 4 * sizeof(float));
856 
857  memmove(&p->samples_in[0], &p->samples_in[AMR_SUBFRAME_SIZE],
858  LP_FILTER_ORDER * sizeof(float));
859 }
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 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  float impulse_buffer[LP_FILTER_ORDER + AMR_TILT_RESPONSE] = { 0 };
880  float *hf = impulse_buffer + LP_FILTER_ORDER; // start of impulse response
881 
882  hf[0] = 1.0;
883  memcpy(hf + 1, lpc_n, sizeof(float) * LP_FILTER_ORDER);
884  p->celpf_ctx.celp_lp_synthesis_filterf(hf, lpc_d, hf,
886  LP_FILTER_ORDER);
887 
888  rh0 = p->celpm_ctx.dot_productf(hf, hf, AMR_TILT_RESPONSE);
889  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  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 static void postfilter(AMRContext *p, float *lpc, float *buf_out)
905 {
906  int i;
907  float *samples = p->samples_in + LP_FILTER_ORDER; // Start of input
908 
909  float speech_gain = p->celpm_ctx.dot_productf(samples, samples,
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  if (p->cur_frame_mode == MODE_12k2 || p->cur_frame_mode == MODE_10k2) {
917  gamma_n = ff_pow_0_7;
918  gamma_d = ff_pow_0_75;
919  } else {
920  gamma_n = ff_pow_0_55;
921  gamma_d = ff_pow_0_7;
922  }
923 
924  for (i = 0; i < LP_FILTER_ORDER; i++) {
925  lpc_n[i] = lpc[i] * gamma_n[i];
926  lpc_d[i] = lpc[i] * gamma_d[i];
927  }
928 
929  memcpy(pole_out, p->postfilter_mem, sizeof(float) * LP_FILTER_ORDER);
930  p->celpf_ctx.celp_lp_synthesis_filterf(pole_out + LP_FILTER_ORDER, lpc_d, samples,
931  AMR_SUBFRAME_SIZE, LP_FILTER_ORDER);
932  memcpy(p->postfilter_mem, pole_out + AMR_SUBFRAME_SIZE,
933  sizeof(float) * LP_FILTER_ORDER);
934 
935  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  ff_tilt_compensation(&p->tilt_mem, tilt_factor(p, lpc_n, lpc_d), buf_out,
941 
942  ff_adaptive_gain_control(buf_out, buf_out, speech_gain, AMR_SUBFRAME_SIZE,
944 }
945 
946 /// @}
947 
948 static int amrnb_decode_frame(AVCodecContext *avctx, void *data,
949  int *got_frame_ptr, AVPacket *avpkt)
950 {
951 
952  AMRContext *p = avctx->priv_data; // pointer to private data
953  AVFrame *frame = data;
954  const uint8_t *buf = avpkt->data;
955  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  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  frame->nb_samples = AMR_BLOCK_SIZE;
966  if ((ret = ff_get_buffer(avctx, frame)) < 0) {
967  av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
968  return ret;
969  }
970  buf_out = (float *)frame->data[0];
971 
972  p->cur_frame_mode = unpack_bitstream(p, buf, buf_size);
973  if (p->cur_frame_mode == NO_DATA) {
974  av_log(avctx, AV_LOG_ERROR, "Corrupt bitstream\n");
975  return AVERROR_INVALIDDATA;
976  }
977  if (p->cur_frame_mode == MODE_DTX) {
978  av_log_missing_feature(avctx, "dtx mode", 0);
979  av_log(avctx, AV_LOG_INFO, "Note: libopencore_amrnb supports dtx\n");
980  return AVERROR_PATCHWELCOME;
981  }
982 
983  if (p->cur_frame_mode == MODE_12k2) {
984  lsf2lsp_5(p);
985  } else
986  lsf2lsp_3(p);
987 
988  for (i = 0; i < 4; i++)
989  ff_acelp_lspd2lpc(p->lsp[i], p->lpc[i], 5);
990 
991  for (subframe = 0; subframe < 4; subframe++) {
992  const AMRNBSubframe *amr_subframe = &p->frame.subframe[subframe];
993 
994  decode_pitch_vector(p, amr_subframe, subframe);
995 
996  decode_fixed_sparse(&fixed_sparse, amr_subframe->pulses,
997  p->cur_frame_mode, subframe);
998 
999  // The fixed gain (section 6.1.3) depends on the fixed vector
1000  // (section 6.1.2), but the fixed vector calculation uses
1001  // pitch sharpening based on the on the pitch gain (section 6.1.3).
1002  // So the correct order is: pitch gain, pitch sharpening, fixed gain.
1003  decode_gains(p, amr_subframe, p->cur_frame_mode, subframe,
1004  &fixed_gain_factor);
1005 
1006  pitch_sharpening(p, subframe, p->cur_frame_mode, &fixed_sparse);
1007 
1008  if (fixed_sparse.pitch_lag == 0) {
1009  av_log(avctx, AV_LOG_ERROR, "The file is corrupted, pitch_lag = 0 is not allowed\n");
1010  return AVERROR_INVALIDDATA;
1011  }
1012  ff_set_fixed_vector(p->fixed_vector, &fixed_sparse, 1.0,
1014 
1015  p->fixed_gain[4] =
1016  ff_amr_set_fixed_gain(fixed_gain_factor,
1018  p->fixed_vector,
1021  p->prediction_error,
1023 
1024  // The excitation feedback is calculated without any processing such
1025  // as fixed gain smoothing. This isn't mentioned in the specification.
1026  for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
1027  p->excitation[i] *= p->pitch_gain[4];
1028  ff_set_fixed_vector(p->excitation, &fixed_sparse, p->fixed_gain[4],
1029  AMR_SUBFRAME_SIZE);
1030 
1031  // In the ref decoder, excitation is stored with no fractional bits.
1032  // This step prevents buzz in silent periods. The ref encoder can
1033  // emit long sequences with pitch factor greater than one. This
1034  // creates unwanted feedback if the excitation vector is nonzero.
1035  // (e.g. test sequence T19_795.COD in 3GPP TS 26.074)
1036  for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
1037  p->excitation[i] = truncf(p->excitation[i]);
1038 
1039  // Smooth fixed gain.
1040  // The specification is ambiguous, but in the reference source, the
1041  // smoothed value is NOT fed back into later fixed gain smoothing.
1042  synth_fixed_gain = fixed_gain_smooth(p, p->lsf_q[subframe],
1043  p->lsf_avg, p->cur_frame_mode);
1044 
1045  synth_fixed_vector = anti_sparseness(p, &fixed_sparse, p->fixed_vector,
1046  synth_fixed_gain, spare_vector);
1047 
1048  if (synthesis(p, p->lpc[subframe], synth_fixed_gain,
1049  synth_fixed_vector, &p->samples_in[LP_FILTER_ORDER], 0))
1050  // overflow detected -> rerun synthesis scaling pitch vector down
1051  // by a factor of 4, skipping pitch vector contribution emphasis
1052  // and adaptive gain control
1053  synthesis(p, p->lpc[subframe], synth_fixed_gain,
1054  synth_fixed_vector, &p->samples_in[LP_FILTER_ORDER], 1);
1055 
1056  postfilter(p, p->lpc[subframe], buf_out + subframe * AMR_SUBFRAME_SIZE);
1057 
1058  // update buffers and history
1059  ff_clear_fixed_vector(p->fixed_vector, &fixed_sparse, AMR_SUBFRAME_SIZE);
1060  update_state(p);
1061  }
1062 
1064  buf_out, highpass_zeros,
1068 
1069  /* Update averaged lsf vector (used for fixed gain smoothing).
1070  *
1071  * Note that lsf_avg should not incorporate the current frame's LSFs
1072  * for fixed_gain_smooth.
1073  * The specification has an incorrect formula: the reference decoder uses
1074  * qbar(n-1) rather than qbar(n) in section 6.1(4) equation 71. */
1076  0.84, 0.16, LP_FILTER_ORDER);
1077 
1078  *got_frame_ptr = 1;
1079 
1080  /* return the amount of bytes consumed if everything was OK */
1081  return frame_sizes_nb[p->cur_frame_mode] + 1; // +7 for rounding and +8 for TOC
1082 }
1083 
1084 
1086  .name = "amrnb",
1087  .type = AVMEDIA_TYPE_AUDIO,
1088  .id = AV_CODEC_ID_AMR_NB,
1089  .priv_data_size = sizeof(AMRContext),
1092  .capabilities = CODEC_CAP_DR1,
1093  .long_name = NULL_IF_CONFIG_SMALL("AMR-NB (Adaptive Multi-Rate NarrowBand)"),
1094  .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT,
1096 };