2 * AMR narrowband decoder
3 * Copyright (c) 2006-2007 Robert Swain
4 * Copyright (c) 2009 Colin McQuillan
6 * This file is part of FFmpeg.
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.
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.
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
26 * AMR narrowband decoder
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.
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.
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
48 #include "libavutil/common.h"
49 #include "celp_math.h"
50 #include "celp_filters.h"
51 #include "acelp_filters.h"
52 #include "acelp_vectors.h"
53 #include "acelp_pitch_delay.h"
56 #include "amrnbdata.h"
58 #define AMR_BLOCK_SIZE 160 ///< samples per frame
59 #define AMR_SAMPLE_BOUND 32768.0 ///< threshold for synthesis overflow
62 * Scale from constructed speech to [-1,1]
64 * AMR is designed to produce 16-bit PCM samples (3GPP TS 26.090 4.2) but
65 * upscales by two (section 6.2.2).
67 * Fundamentally, this scale is determined by energy_mean through
68 * the fixed vector contribution to the excitation vector.
70 #define AMR_SAMPLE_SCALE (2.0 / 32768.0)
72 /** Prediction factor for 12.2kbit/s mode */
73 #define PRED_FAC_MODE_12k2 0.65
75 #define LSF_R_FAC (8000.0 / 32768.0) ///< LSF residual tables to Hertz
76 #define MIN_LSF_SPACING (50.0488 / 8000.0) ///< Ensures stability of LPC filter
77 #define PITCH_LAG_MIN_MODE_12k2 18 ///< Lower bound on decoded lag search in 12.2kbit/s mode
79 /** Initial energy in dB. Also used for bad frames (unimplemented). */
80 #define MIN_ENERGY -14.0
82 /** Maximum sharpening factor
84 * The specification says 0.8, which should be 13107, but the reference C code
85 * uses 13017 instead. (Amusingly the same applies to SHARP_MAX in g729dec.c.)
87 #define SHARP_MAX 0.79449462890625
89 /** Number of impulse response coefficients used for tilt factor */
90 #define AMR_TILT_RESPONSE 22
91 /** Tilt factor = 1st reflection coefficient * gamma_t */
92 #define AMR_TILT_GAMMA_T 0.8
93 /** Adaptive gain control factor used in post-filter */
94 #define AMR_AGC_ALPHA 0.9
96 typedef struct AMRContext {
97 AMRNBFrame frame; ///< decoded AMR parameters (lsf coefficients, codebook indexes, etc)
98 uint8_t bad_frame_indicator; ///< bad frame ? 1 : 0
99 enum Mode cur_frame_mode;
101 int16_t prev_lsf_r[LP_FILTER_ORDER]; ///< residual LSF vector from previous subframe
102 double lsp[4][LP_FILTER_ORDER]; ///< lsp vectors from current frame
103 double prev_lsp_sub4[LP_FILTER_ORDER]; ///< lsp vector for the 4th subframe of the previous frame
105 float lsf_q[4][LP_FILTER_ORDER]; ///< Interpolated LSF vector for fixed gain smoothing
106 float lsf_avg[LP_FILTER_ORDER]; ///< vector of averaged lsf vector
108 float lpc[4][LP_FILTER_ORDER]; ///< lpc coefficient vectors for 4 subframes
110 uint8_t pitch_lag_int; ///< integer part of pitch lag from current subframe
112 float excitation_buf[PITCH_DELAY_MAX + LP_FILTER_ORDER + 1 + AMR_SUBFRAME_SIZE]; ///< current excitation and all necessary excitation history
113 float *excitation; ///< pointer to the current excitation vector in excitation_buf
115 float pitch_vector[AMR_SUBFRAME_SIZE]; ///< adaptive code book (pitch) vector
116 float fixed_vector[AMR_SUBFRAME_SIZE]; ///< algebraic codebook (fixed) vector (must be kept zero between frames)
118 float prediction_error[4]; ///< quantified prediction errors {20log10(^gamma_gc)} for previous four subframes
119 float pitch_gain[5]; ///< quantified pitch gains for the current and previous four subframes
120 float fixed_gain[5]; ///< quantified fixed gains for the current and previous four subframes
122 float beta; ///< previous pitch_gain, bounded by [0.0,SHARP_MAX]
123 uint8_t diff_count; ///< the number of subframes for which diff has been above 0.65
124 uint8_t hang_count; ///< the number of subframes since a hangover period started
126 float prev_sparse_fixed_gain; ///< previous fixed gain; used by anti-sparseness processing to determine "onset"
127 uint8_t prev_ir_filter_nr; ///< previous impulse response filter "impNr": 0 - strong, 1 - medium, 2 - none
128 uint8_t ir_filter_onset; ///< flag for impulse response filter strength
130 float postfilter_mem[10]; ///< previous intermediate values in the formant filter
131 float tilt_mem; ///< previous input to tilt compensation filter
132 float postfilter_agc; ///< previous factor used for adaptive gain control
133 float high_pass_mem[2]; ///< previous intermediate values in the high-pass filter
135 float samples_in[LP_FILTER_ORDER + AMR_SUBFRAME_SIZE]; ///< floating point samples
139 /** Double version of ff_weighted_vector_sumf() */
140 static void weighted_vector_sumd(double *out, const double *in_a,
141 const double *in_b, double weight_coeff_a,
142 double weight_coeff_b, int length)
146 for (i = 0; i < length; i++)
147 out[i] = weight_coeff_a * in_a[i]
148 + weight_coeff_b * in_b[i];
151 static av_cold int amrnb_decode_init(AVCodecContext *avctx)
153 AMRContext *p = avctx->priv_data;
156 avctx->sample_fmt = SAMPLE_FMT_FLT;
158 // p->excitation always points to the same position in p->excitation_buf
159 p->excitation = &p->excitation_buf[PITCH_DELAY_MAX + LP_FILTER_ORDER + 1];
161 for (i = 0; i < LP_FILTER_ORDER; i++) {
162 p->prev_lsp_sub4[i] = lsp_sub4_init[i] * 1000 / (float)(1 << 15);
163 p->lsf_avg[i] = p->lsf_q[3][i] = lsp_avg_init[i] / (float)(1 << 15);
166 for (i = 0; i < 4; i++)
167 p->prediction_error[i] = MIN_ENERGY;
174 * Unpack an RFC4867 speech frame into the AMR frame mode and parameters.
176 * The order of speech bits is specified by 3GPP TS 26.101.
178 * @param p the context
179 * @param buf pointer to the input buffer
180 * @param buf_size size of the input buffer
182 * @return the frame mode
184 static enum Mode unpack_bitstream(AMRContext *p, const uint8_t *buf,
190 init_get_bits(&gb, buf, buf_size * 8);
192 // Decode the first octet.
193 skip_bits(&gb, 1); // padding bit
194 mode = get_bits(&gb, 4); // frame type
195 p->bad_frame_indicator = !get_bits1(&gb); // quality bit
196 skip_bits(&gb, 2); // two padding bits
198 if (mode < MODE_DTX) {
199 uint16_t *data = (uint16_t *)&p->frame;
200 const uint8_t *order = amr_unpacking_bitmaps_per_mode[mode];
203 memset(&p->frame, 0, sizeof(AMRNBFrame));
205 while ((field_size = *order++)) {
207 int field_offset = *order++;
208 while (field_size--) {
211 field |= buf[bit >> 3] >> (bit & 7) & 1;
213 data[field_offset] = field;
221 /// @defgroup amr_lpc_decoding AMR pitch LPC coefficient decoding functions
225 * Convert an lsf vector into an lsp vector.
227 * @param lsf input lsf vector
228 * @param lsp output lsp vector
230 static void lsf2lsp(const float *lsf, double *lsp)
234 for (i = 0; i < LP_FILTER_ORDER; i++)
235 lsp[i] = cos(2.0 * M_PI * lsf[i]);
239 * Interpolate the LSF vector (used for fixed gain smoothing).
240 * The interpolation is done over all four subframes even in MODE_12k2.
242 * @param[in,out] lsf_q LSFs in [0,1] for each subframe
243 * @param[in] lsf_new New LSFs in [0,1] for subframe 4
245 static void interpolate_lsf(float lsf_q[4][LP_FILTER_ORDER], float *lsf_new)
249 for (i = 0; i < 4; i++)
250 ff_weighted_vector_sumf(lsf_q[i], lsf_q[3], lsf_new,
251 0.25 * (3 - i), 0.25 * (i + 1),
256 * Decode a set of 5 split-matrix quantized lsf indexes into an lsp vector.
258 * @param p the context
259 * @param lsp output LSP vector
260 * @param lsf_no_r LSF vector without the residual vector added
261 * @param lsf_quantizer pointers to LSF dictionary tables
262 * @param quantizer_offset offset in tables
263 * @param sign for the 3 dictionary table
264 * @param update store data for computing the next frame's LSFs
266 static void lsf2lsp_for_mode12k2(AMRContext *p, double lsp[LP_FILTER_ORDER],
267 const float lsf_no_r[LP_FILTER_ORDER],
268 const int16_t *lsf_quantizer[5],
269 const int quantizer_offset,
270 const int sign, const int update)
272 int16_t lsf_r[LP_FILTER_ORDER]; // residual LSF vector
273 float lsf_q[LP_FILTER_ORDER]; // quantified LSF vector
276 for (i = 0; i < LP_FILTER_ORDER >> 1; i++)
277 memcpy(&lsf_r[i << 1], &lsf_quantizer[i][quantizer_offset],
286 memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(float));
288 for (i = 0; i < LP_FILTER_ORDER; i++)
289 lsf_q[i] = lsf_r[i] * (LSF_R_FAC / 8000.0) + lsf_no_r[i] * (1.0 / 8000.0);
291 ff_set_min_dist_lsf(lsf_q, MIN_LSF_SPACING, LP_FILTER_ORDER);
294 interpolate_lsf(p->lsf_q, lsf_q);
300 * Decode a set of 5 split-matrix quantized lsf indexes into 2 lsp vectors.
302 * @param p pointer to the AMRContext
304 static void lsf2lsp_5(AMRContext *p)
306 const uint16_t *lsf_param = p->frame.lsf;
307 float lsf_no_r[LP_FILTER_ORDER]; // LSFs without the residual vector
308 const int16_t *lsf_quantizer[5];
311 lsf_quantizer[0] = lsf_5_1[lsf_param[0]];
312 lsf_quantizer[1] = lsf_5_2[lsf_param[1]];
313 lsf_quantizer[2] = lsf_5_3[lsf_param[2] >> 1];
314 lsf_quantizer[3] = lsf_5_4[lsf_param[3]];
315 lsf_quantizer[4] = lsf_5_5[lsf_param[4]];
317 for (i = 0; i < LP_FILTER_ORDER; i++)
318 lsf_no_r[i] = p->prev_lsf_r[i] * LSF_R_FAC * PRED_FAC_MODE_12k2 + lsf_5_mean[i];
320 lsf2lsp_for_mode12k2(p, p->lsp[1], lsf_no_r, lsf_quantizer, 0, lsf_param[2] & 1, 0);
321 lsf2lsp_for_mode12k2(p, p->lsp[3], lsf_no_r, lsf_quantizer, 2, lsf_param[2] & 1, 1);
323 // interpolate LSP vectors at subframes 1 and 3
324 weighted_vector_sumd(p->lsp[0], p->prev_lsp_sub4, p->lsp[1], 0.5, 0.5, LP_FILTER_ORDER);
325 weighted_vector_sumd(p->lsp[2], p->lsp[1] , p->lsp[3], 0.5, 0.5, LP_FILTER_ORDER);
329 * Decode a set of 3 split-matrix quantized lsf indexes into an lsp vector.
331 * @param p pointer to the AMRContext
333 static void lsf2lsp_3(AMRContext *p)
335 const uint16_t *lsf_param = p->frame.lsf;
336 int16_t lsf_r[LP_FILTER_ORDER]; // residual LSF vector
337 float lsf_q[LP_FILTER_ORDER]; // quantified LSF vector
338 const int16_t *lsf_quantizer;
341 lsf_quantizer = (p->cur_frame_mode == MODE_7k95 ? lsf_3_1_MODE_7k95 : lsf_3_1)[lsf_param[0]];
342 memcpy(lsf_r, lsf_quantizer, 3 * sizeof(*lsf_r));
344 lsf_quantizer = lsf_3_2[lsf_param[1] << (p->cur_frame_mode <= MODE_5k15)];
345 memcpy(lsf_r + 3, lsf_quantizer, 3 * sizeof(*lsf_r));
347 lsf_quantizer = (p->cur_frame_mode <= MODE_5k15 ? lsf_3_3_MODE_5k15 : lsf_3_3)[lsf_param[2]];
348 memcpy(lsf_r + 6, lsf_quantizer, 4 * sizeof(*lsf_r));
350 // calculate mean-removed LSF vector and add mean
351 for (i = 0; i < LP_FILTER_ORDER; i++)
352 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);
354 ff_set_min_dist_lsf(lsf_q, MIN_LSF_SPACING, LP_FILTER_ORDER);
356 // store data for computing the next frame's LSFs
357 interpolate_lsf(p->lsf_q, lsf_q);
358 memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(*lsf_r));
360 lsf2lsp(lsf_q, p->lsp[3]);
362 // interpolate LSP vectors at subframes 1, 2 and 3
363 for (i = 1; i <= 3; i++)
364 for(j = 0; j < LP_FILTER_ORDER; j++)
365 p->lsp[i-1][j] = p->prev_lsp_sub4[j] +
366 (p->lsp[3][j] - p->prev_lsp_sub4[j]) * 0.25 * i;
372 /// @defgroup amr_pitch_vector_decoding AMR pitch vector decoding functions
376 * Like ff_decode_pitch_lag(), but with 1/6 resolution
378 static void decode_pitch_lag_1_6(int *lag_int, int *lag_frac, int pitch_index,
379 const int prev_lag_int, const int subframe)
381 if (subframe == 0 || subframe == 2) {
382 if (pitch_index < 463) {
383 *lag_int = (pitch_index + 107) * 10923 >> 16;
384 *lag_frac = pitch_index - *lag_int * 6 + 105;
386 *lag_int = pitch_index - 368;
390 *lag_int = ((pitch_index + 5) * 10923 >> 16) - 1;
391 *lag_frac = pitch_index - *lag_int * 6 - 3;
392 *lag_int += av_clip(prev_lag_int - 5, PITCH_LAG_MIN_MODE_12k2,
393 PITCH_DELAY_MAX - 9);
397 static void decode_pitch_vector(AMRContext *p,
398 const AMRNBSubframe *amr_subframe,
401 int pitch_lag_int, pitch_lag_frac;
402 enum Mode mode = p->cur_frame_mode;
404 if (p->cur_frame_mode == MODE_12k2) {
405 decode_pitch_lag_1_6(&pitch_lag_int, &pitch_lag_frac,
406 amr_subframe->p_lag, p->pitch_lag_int,
409 ff_decode_pitch_lag(&pitch_lag_int, &pitch_lag_frac,
411 p->pitch_lag_int, subframe,
412 mode != MODE_4k75 && mode != MODE_5k15,
413 mode <= MODE_6k7 ? 4 : (mode == MODE_7k95 ? 5 : 6));
415 p->pitch_lag_int = pitch_lag_int; // store previous lag in a uint8_t
417 pitch_lag_frac <<= (p->cur_frame_mode != MODE_12k2);
419 pitch_lag_int += pitch_lag_frac > 0;
421 /* Calculate the pitch vector by interpolating the past excitation at the
422 pitch lag using a b60 hamming windowed sinc function. */
423 ff_acelp_interpolatef(p->excitation, p->excitation + 1 - pitch_lag_int,
425 pitch_lag_frac + 6 - 6*(pitch_lag_frac > 0),
426 10, AMR_SUBFRAME_SIZE);
428 memcpy(p->pitch_vector, p->excitation, AMR_SUBFRAME_SIZE * sizeof(float));
434 /// @defgroup amr_algebraic_code_book AMR algebraic code book (fixed) vector decoding functions
438 * Decode a 10-bit algebraic codebook index from a 10.2 kbit/s frame.
440 static void decode_10bit_pulse(int code, int pulse_position[8],
441 int i1, int i2, int i3)
443 // coded using 7+3 bits with the 3 LSBs being, individually, the LSB of 1 of
444 // the 3 pulses and the upper 7 bits being coded in base 5
445 const uint8_t *positions = base_five_table[code >> 3];
446 pulse_position[i1] = (positions[2] << 1) + ( code & 1);
447 pulse_position[i2] = (positions[1] << 1) + ((code >> 1) & 1);
448 pulse_position[i3] = (positions[0] << 1) + ((code >> 2) & 1);
452 * Decode the algebraic codebook index to pulse positions and signs and
453 * construct the algebraic codebook vector for MODE_10k2.
455 * @param fixed_index positions of the eight pulses
456 * @param fixed_sparse pointer to the algebraic codebook vector
458 static void decode_8_pulses_31bits(const int16_t *fixed_index,
459 AMRFixed *fixed_sparse)
461 int pulse_position[8];
464 decode_10bit_pulse(fixed_index[4], pulse_position, 0, 4, 1);
465 decode_10bit_pulse(fixed_index[5], pulse_position, 2, 6, 5);
467 // coded using 5+2 bits with the 2 LSBs being, individually, the LSB of 1 of
468 // the 2 pulses and the upper 5 bits being coded in base 5
469 temp = ((fixed_index[6] >> 2) * 25 + 12) >> 5;
470 pulse_position[3] = temp % 5;
471 pulse_position[7] = temp / 5;
472 if (pulse_position[7] & 1)
473 pulse_position[3] = 4 - pulse_position[3];
474 pulse_position[3] = (pulse_position[3] << 1) + ( fixed_index[6] & 1);
475 pulse_position[7] = (pulse_position[7] << 1) + ((fixed_index[6] >> 1) & 1);
478 for (i = 0; i < 4; i++) {
479 const int pos1 = (pulse_position[i] << 2) + i;
480 const int pos2 = (pulse_position[i + 4] << 2) + i;
481 const float sign = fixed_index[i] ? -1.0 : 1.0;
482 fixed_sparse->x[i ] = pos1;
483 fixed_sparse->x[i + 4] = pos2;
484 fixed_sparse->y[i ] = sign;
485 fixed_sparse->y[i + 4] = pos2 < pos1 ? -sign : sign;
490 * Decode the algebraic codebook index to pulse positions and signs,
491 * then construct the algebraic codebook vector.
493 * nb of pulses | bits encoding pulses
494 * For MODE_4k75 or MODE_5k15, 2 | 1-3, 4-6, 7
495 * MODE_5k9, 2 | 1, 2-4, 5-6, 7-9
496 * MODE_6k7, 3 | 1-3, 4, 5-7, 8, 9-11
497 * MODE_7k4 or MODE_7k95, 4 | 1-3, 4-6, 7-9, 10, 11-13
499 * @param fixed_sparse pointer to the algebraic codebook vector
500 * @param pulses algebraic codebook indexes
501 * @param mode mode of the current frame
502 * @param subframe current subframe number
504 static void decode_fixed_sparse(AMRFixed *fixed_sparse, const uint16_t *pulses,
505 const enum Mode mode, const int subframe)
507 assert(MODE_4k75 <= mode && mode <= MODE_12k2);
509 if (mode == MODE_12k2) {
510 ff_decode_10_pulses_35bits(pulses, fixed_sparse, gray_decode, 5, 3);
511 } else if (mode == MODE_10k2) {
512 decode_8_pulses_31bits(pulses, fixed_sparse);
514 int *pulse_position = fixed_sparse->x;
516 const int fixed_index = pulses[0];
518 if (mode <= MODE_5k15) {
519 pulse_subset = ((fixed_index >> 3) & 8) + (subframe << 1);
520 pulse_position[0] = ( fixed_index & 7) * 5 + track_position[pulse_subset];
521 pulse_position[1] = ((fixed_index >> 3) & 7) * 5 + track_position[pulse_subset + 1];
523 } else if (mode == MODE_5k9) {
524 pulse_subset = ((fixed_index & 1) << 1) + 1;
525 pulse_position[0] = ((fixed_index >> 1) & 7) * 5 + pulse_subset;
526 pulse_subset = (fixed_index >> 4) & 3;
527 pulse_position[1] = ((fixed_index >> 6) & 7) * 5 + pulse_subset + (pulse_subset == 3 ? 1 : 0);
528 fixed_sparse->n = pulse_position[0] == pulse_position[1] ? 1 : 2;
529 } else if (mode == MODE_6k7) {
530 pulse_position[0] = (fixed_index & 7) * 5;
531 pulse_subset = (fixed_index >> 2) & 2;
532 pulse_position[1] = ((fixed_index >> 4) & 7) * 5 + pulse_subset + 1;
533 pulse_subset = (fixed_index >> 6) & 2;
534 pulse_position[2] = ((fixed_index >> 8) & 7) * 5 + pulse_subset + 2;
536 } else { // mode <= MODE_7k95
537 pulse_position[0] = gray_decode[ fixed_index & 7];
538 pulse_position[1] = gray_decode[(fixed_index >> 3) & 7] + 1;
539 pulse_position[2] = gray_decode[(fixed_index >> 6) & 7] + 2;
540 pulse_subset = (fixed_index >> 9) & 1;
541 pulse_position[3] = gray_decode[(fixed_index >> 10) & 7] + pulse_subset + 3;
544 for (i = 0; i < fixed_sparse->n; i++)
545 fixed_sparse->y[i] = (pulses[1] >> i) & 1 ? 1.0 : -1.0;
550 * Apply pitch lag to obtain the sharpened fixed vector (section 6.1.2)
552 * @param p the context
553 * @param subframe unpacked amr subframe
554 * @param mode mode of the current frame
555 * @param fixed_sparse sparse respresentation of the fixed vector
557 static void pitch_sharpening(AMRContext *p, int subframe, enum Mode mode,
558 AMRFixed *fixed_sparse)
560 // The spec suggests the current pitch gain is always used, but in other
561 // modes the pitch and codebook gains are joinly quantized (sec 5.8.2)
562 // so the codebook gain cannot depend on the quantized pitch gain.
563 if (mode == MODE_12k2)
564 p->beta = FFMIN(p->pitch_gain[4], 1.0);
566 fixed_sparse->pitch_lag = p->pitch_lag_int;
567 fixed_sparse->pitch_fac = p->beta;
569 // Save pitch sharpening factor for the next subframe
570 // MODE_4k75 only updates on the 2nd and 4th subframes - this follows from
571 // the fact that the gains for two subframes are jointly quantized.
572 if (mode != MODE_4k75 || subframe & 1)
573 p->beta = av_clipf(p->pitch_gain[4], 0.0, SHARP_MAX);
578 /// @defgroup amr_gain_decoding AMR gain decoding functions
582 * fixed gain smoothing
583 * Note that where the spec specifies the "spectrum in the q domain"
584 * in section 6.1.4, in fact frequencies should be used.
586 * @param p the context
587 * @param lsf LSFs for the current subframe, in the range [0,1]
588 * @param lsf_avg averaged LSFs
589 * @param mode mode of the current frame
591 * @return fixed gain smoothed
593 static float fixed_gain_smooth(AMRContext *p , const float *lsf,
594 const float *lsf_avg, const enum Mode mode)
599 for (i = 0; i < LP_FILTER_ORDER; i++)
600 diff += fabs(lsf_avg[i] - lsf[i]) / lsf_avg[i];
602 // If diff is large for ten subframes, disable smoothing for a 40-subframe
608 if (p->diff_count > 10) {
610 p->diff_count--; // don't let diff_count overflow
613 if (p->hang_count < 40) {
615 } else if (mode < MODE_7k4 || mode == MODE_10k2) {
616 const float smoothing_factor = av_clipf(4.0 * diff - 1.6, 0.0, 1.0);
617 const float fixed_gain_mean = (p->fixed_gain[0] + p->fixed_gain[1] +
618 p->fixed_gain[2] + p->fixed_gain[3] +
619 p->fixed_gain[4]) * 0.2;
620 return smoothing_factor * p->fixed_gain[4] +
621 (1.0 - smoothing_factor) * fixed_gain_mean;
623 return p->fixed_gain[4];
627 * Decode pitch gain and fixed gain factor (part of section 6.1.3).
629 * @param p the context
630 * @param amr_subframe unpacked amr subframe
631 * @param mode mode of the current frame
632 * @param subframe current subframe number
633 * @param fixed_gain_factor decoded gain correction factor
635 static void decode_gains(AMRContext *p, const AMRNBSubframe *amr_subframe,
636 const enum Mode mode, const int subframe,
637 float *fixed_gain_factor)
639 if (mode == MODE_12k2 || mode == MODE_7k95) {
640 p->pitch_gain[4] = qua_gain_pit [amr_subframe->p_gain ]
642 *fixed_gain_factor = qua_gain_code[amr_subframe->fixed_gain]
645 const uint16_t *gains;
647 if (mode >= MODE_6k7) {
648 gains = gains_high[amr_subframe->p_gain];
649 } else if (mode >= MODE_5k15) {
650 gains = gains_low [amr_subframe->p_gain];
652 // gain index is only coded in subframes 0,2 for MODE_4k75
653 gains = gains_MODE_4k75[(p->frame.subframe[subframe & 2].p_gain << 1) + (subframe & 1)];
656 p->pitch_gain[4] = gains[0] * (1.0 / 16384.0);
657 *fixed_gain_factor = gains[1] * (1.0 / 4096.0);
664 /// @defgroup amr_pre_processing AMR pre-processing functions
668 * Circularly convolve a sparse fixed vector with a phase dispersion impulse
669 * response filter (D.6.2 of G.729 and 6.1.5 of AMR).
671 * @param out vector with filter applied
672 * @param in source vector
673 * @param filter phase filter coefficients
675 * out[n] = sum(i,0,len-1){ in[i] * filter[(len + n - i)%len] }
677 static void apply_ir_filter(float *out, const AMRFixed *in,
680 float filter1[AMR_SUBFRAME_SIZE], //!< filters at pitch lag*1 and *2
681 filter2[AMR_SUBFRAME_SIZE];
682 int lag = in->pitch_lag;
683 float fac = in->pitch_fac;
686 if (lag < AMR_SUBFRAME_SIZE) {
687 ff_celp_circ_addf(filter1, filter, filter, lag, fac,
690 if (lag < AMR_SUBFRAME_SIZE >> 1)
691 ff_celp_circ_addf(filter2, filter, filter1, lag, fac,
695 memset(out, 0, sizeof(float) * AMR_SUBFRAME_SIZE);
696 for (i = 0; i < in->n; i++) {
699 const float *filterp;
701 if (x >= AMR_SUBFRAME_SIZE - lag) {
703 } else if (x >= AMR_SUBFRAME_SIZE - (lag << 1)) {
708 ff_celp_circ_addf(out, out, filterp, x, y, AMR_SUBFRAME_SIZE);
713 * Reduce fixed vector sparseness by smoothing with one of three IR filters.
714 * Also know as "adaptive phase dispersion".
716 * This implements 3GPP TS 26.090 section 6.1(5).
718 * @param p the context
719 * @param fixed_sparse algebraic codebook vector
720 * @param fixed_vector unfiltered fixed vector
721 * @param fixed_gain smoothed gain
722 * @param out space for modified vector if necessary
724 static const float *anti_sparseness(AMRContext *p, AMRFixed *fixed_sparse,
725 const float *fixed_vector,
726 float fixed_gain, float *out)
730 if (p->pitch_gain[4] < 0.6) {
731 ir_filter_nr = 0; // strong filtering
732 } else if (p->pitch_gain[4] < 0.9) {
733 ir_filter_nr = 1; // medium filtering
735 ir_filter_nr = 2; // no filtering
738 if (fixed_gain > 2.0 * p->prev_sparse_fixed_gain) {
739 p->ir_filter_onset = 2;
740 } else if (p->ir_filter_onset)
741 p->ir_filter_onset--;
743 if (!p->ir_filter_onset) {
746 for (i = 0; i < 5; i++)
747 if (p->pitch_gain[i] < 0.6)
752 if (ir_filter_nr > p->prev_ir_filter_nr + 1)
754 } else if (ir_filter_nr < 2)
757 // Disable filtering for very low level of fixed_gain.
758 // Note this step is not specified in the technical description but is in
759 // the reference source in the function Ph_disp.
760 if (fixed_gain < 5.0)
763 if (p->cur_frame_mode != MODE_7k4 && p->cur_frame_mode < MODE_10k2
764 && ir_filter_nr < 2) {
765 apply_ir_filter(out, fixed_sparse,
766 (p->cur_frame_mode == MODE_7k95 ?
767 ir_filters_lookup_MODE_7k95 :
768 ir_filters_lookup)[ir_filter_nr]);
772 // update ir filter strength history
773 p->prev_ir_filter_nr = ir_filter_nr;
774 p->prev_sparse_fixed_gain = fixed_gain;
782 /// @defgroup amr_synthesis AMR synthesis functions
786 * Conduct 10th order linear predictive coding synthesis.
788 * @param p pointer to the AMRContext
789 * @param lpc pointer to the LPC coefficients
790 * @param fixed_gain fixed codebook gain for synthesis
791 * @param fixed_vector algebraic codebook vector
792 * @param samples pointer to the output speech samples
793 * @param overflow 16-bit overflow flag
795 static int synthesis(AMRContext *p, float *lpc,
796 float fixed_gain, const float *fixed_vector,
797 float *samples, uint8_t overflow)
800 float excitation[AMR_SUBFRAME_SIZE];
802 // if an overflow has been detected, the pitch vector is scaled down by a
805 for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
806 p->pitch_vector[i] *= 0.25;
808 ff_weighted_vector_sumf(excitation, p->pitch_vector, fixed_vector,
809 p->pitch_gain[4], fixed_gain, AMR_SUBFRAME_SIZE);
811 // emphasize pitch vector contribution
812 if (p->pitch_gain[4] > 0.5 && !overflow) {
813 float energy = ff_dot_productf(excitation, excitation,
817 (p->cur_frame_mode == MODE_12k2 ?
818 0.25 * FFMIN(p->pitch_gain[4], 1.0) :
819 0.5 * FFMIN(p->pitch_gain[4], SHARP_MAX));
821 for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
822 excitation[i] += pitch_factor * p->pitch_vector[i];
824 ff_scale_vector_to_given_sum_of_squares(excitation, excitation, energy,
828 ff_celp_lp_synthesis_filterf(samples, lpc, excitation, AMR_SUBFRAME_SIZE,
832 for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
833 if (fabsf(samples[i]) > AMR_SAMPLE_BOUND) {
843 /// @defgroup amr_update AMR update functions
847 * Update buffers and history at the end of decoding a subframe.
849 * @param p pointer to the AMRContext
851 static void update_state(AMRContext *p)
853 memcpy(p->prev_lsp_sub4, p->lsp[3], LP_FILTER_ORDER * sizeof(p->lsp[3][0]));
855 memmove(&p->excitation_buf[0], &p->excitation_buf[AMR_SUBFRAME_SIZE],
856 (PITCH_DELAY_MAX + LP_FILTER_ORDER + 1) * sizeof(float));
858 memmove(&p->pitch_gain[0], &p->pitch_gain[1], 4 * sizeof(float));
859 memmove(&p->fixed_gain[0], &p->fixed_gain[1], 4 * sizeof(float));
861 memmove(&p->samples_in[0], &p->samples_in[AMR_SUBFRAME_SIZE],
862 LP_FILTER_ORDER * sizeof(float));
868 /// @defgroup amr_postproc AMR Post processing functions
872 * Get the tilt factor of a formant filter from its transfer function
874 * @param lpc_n LP_FILTER_ORDER coefficients of the numerator
875 * @param lpc_d LP_FILTER_ORDER coefficients of the denominator
877 static float tilt_factor(float *lpc_n, float *lpc_d)
879 float rh0, rh1; // autocorrelation at lag 0 and 1
881 // LP_FILTER_ORDER prior zeros are needed for ff_celp_lp_synthesis_filterf
882 float impulse_buffer[LP_FILTER_ORDER + AMR_TILT_RESPONSE] = { 0 };
883 float *hf = impulse_buffer + LP_FILTER_ORDER; // start of impulse response
886 memcpy(hf + 1, lpc_n, sizeof(float) * LP_FILTER_ORDER);
887 ff_celp_lp_synthesis_filterf(hf, lpc_d, hf, AMR_TILT_RESPONSE,
890 rh0 = ff_dot_productf(hf, hf, AMR_TILT_RESPONSE);
891 rh1 = ff_dot_productf(hf, hf + 1, AMR_TILT_RESPONSE - 1);
893 // The spec only specifies this check for 12.2 and 10.2 kbit/s
894 // modes. But in the ref source the tilt is always non-negative.
895 return rh1 >= 0.0 ? rh1 / rh0 * AMR_TILT_GAMMA_T : 0.0;
899 * Perform adaptive post-filtering to enhance the quality of the speech.
902 * @param p pointer to the AMRContext
903 * @param lpc interpolated LP coefficients for this subframe
904 * @param buf_out output of the filter
906 static void postfilter(AMRContext *p, float *lpc, float *buf_out)
909 float *samples = p->samples_in + LP_FILTER_ORDER; // Start of input
911 float speech_gain = ff_dot_productf(samples, samples,
914 float pole_out[AMR_SUBFRAME_SIZE + LP_FILTER_ORDER]; // Output of pole filter
915 const float *gamma_n, *gamma_d; // Formant filter factor table
916 float lpc_n[LP_FILTER_ORDER], lpc_d[LP_FILTER_ORDER]; // Transfer function coefficients
918 if (p->cur_frame_mode == MODE_12k2 || p->cur_frame_mode == MODE_10k2) {
919 gamma_n = ff_pow_0_7;
920 gamma_d = ff_pow_0_75;
922 gamma_n = ff_pow_0_55;
923 gamma_d = ff_pow_0_7;
926 for (i = 0; i < LP_FILTER_ORDER; i++) {
927 lpc_n[i] = lpc[i] * gamma_n[i];
928 lpc_d[i] = lpc[i] * gamma_d[i];
931 memcpy(pole_out, p->postfilter_mem, sizeof(float) * LP_FILTER_ORDER);
932 ff_celp_lp_synthesis_filterf(pole_out + LP_FILTER_ORDER, lpc_d, samples,
933 AMR_SUBFRAME_SIZE, LP_FILTER_ORDER);
934 memcpy(p->postfilter_mem, pole_out + AMR_SUBFRAME_SIZE,
935 sizeof(float) * LP_FILTER_ORDER);
937 ff_celp_lp_zero_synthesis_filterf(buf_out, lpc_n,
938 pole_out + LP_FILTER_ORDER,
939 AMR_SUBFRAME_SIZE, LP_FILTER_ORDER);
941 ff_tilt_compensation(&p->tilt_mem, tilt_factor(lpc_n, lpc_d), buf_out,
944 ff_adaptive_gain_control(buf_out, buf_out, speech_gain, AMR_SUBFRAME_SIZE,
945 AMR_AGC_ALPHA, &p->postfilter_agc);
950 static int amrnb_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
954 AMRContext *p = avctx->priv_data; // pointer to private data
955 const uint8_t *buf = avpkt->data;
956 int buf_size = avpkt->size;
957 float *buf_out = data; // pointer to the output data buffer
959 float fixed_gain_factor;
960 AMRFixed fixed_sparse = {0}; // fixed vector up to anti-sparseness processing
961 float spare_vector[AMR_SUBFRAME_SIZE]; // extra stack space to hold result from anti-sparseness processing
962 float synth_fixed_gain; // the fixed gain that synthesis should use
963 const float *synth_fixed_vector; // pointer to the fixed vector that synthesis should use
965 p->cur_frame_mode = unpack_bitstream(p, buf, buf_size);
966 if (p->cur_frame_mode == MODE_DTX) {
967 av_log_missing_feature(avctx, "dtx mode", 1);
971 if (p->cur_frame_mode == MODE_12k2) {
976 for (i = 0; i < 4; i++)
977 ff_acelp_lspd2lpc(p->lsp[i], p->lpc[i], 5);
979 for (subframe = 0; subframe < 4; subframe++) {
980 const AMRNBSubframe *amr_subframe = &p->frame.subframe[subframe];
982 decode_pitch_vector(p, amr_subframe, subframe);
984 decode_fixed_sparse(&fixed_sparse, amr_subframe->pulses,
985 p->cur_frame_mode, subframe);
987 // The fixed gain (section 6.1.3) depends on the fixed vector
988 // (section 6.1.2), but the fixed vector calculation uses
989 // pitch sharpening based on the on the pitch gain (section 6.1.3).
990 // So the correct order is: pitch gain, pitch sharpening, fixed gain.
991 decode_gains(p, amr_subframe, p->cur_frame_mode, subframe,
994 pitch_sharpening(p, subframe, p->cur_frame_mode, &fixed_sparse);
996 ff_set_fixed_vector(p->fixed_vector, &fixed_sparse, 1.0,
1000 ff_amr_set_fixed_gain(fixed_gain_factor,
1001 ff_dot_productf(p->fixed_vector, p->fixed_vector,
1002 AMR_SUBFRAME_SIZE)/AMR_SUBFRAME_SIZE,
1003 p->prediction_error,
1004 energy_mean[p->cur_frame_mode], energy_pred_fac);
1006 // The excitation feedback is calculated without any processing such
1007 // as fixed gain smoothing. This isn't mentioned in the specification.
1008 for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
1009 p->excitation[i] *= p->pitch_gain[4];
1010 ff_set_fixed_vector(p->excitation, &fixed_sparse, p->fixed_gain[4],
1013 // In the ref decoder, excitation is stored with no fractional bits.
1014 // This step prevents buzz in silent periods. The ref encoder can
1015 // emit long sequences with pitch factor greater than one. This
1016 // creates unwanted feedback if the excitation vector is nonzero.
1017 // (e.g. test sequence T19_795.COD in 3GPP TS 26.074)
1018 for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
1019 p->excitation[i] = truncf(p->excitation[i]);
1021 // Smooth fixed gain.
1022 // The specification is ambiguous, but in the reference source, the
1023 // smoothed value is NOT fed back into later fixed gain smoothing.
1024 synth_fixed_gain = fixed_gain_smooth(p, p->lsf_q[subframe],
1025 p->lsf_avg, p->cur_frame_mode);
1027 synth_fixed_vector = anti_sparseness(p, &fixed_sparse, p->fixed_vector,
1028 synth_fixed_gain, spare_vector);
1030 if (synthesis(p, p->lpc[subframe], synth_fixed_gain,
1031 synth_fixed_vector, &p->samples_in[LP_FILTER_ORDER], 0))
1032 // overflow detected -> rerun synthesis scaling pitch vector down
1033 // by a factor of 4, skipping pitch vector contribution emphasis
1034 // and adaptive gain control
1035 synthesis(p, p->lpc[subframe], synth_fixed_gain,
1036 synth_fixed_vector, &p->samples_in[LP_FILTER_ORDER], 1);
1038 postfilter(p, p->lpc[subframe], buf_out + subframe * AMR_SUBFRAME_SIZE);
1040 // update buffers and history
1041 ff_clear_fixed_vector(p->fixed_vector, &fixed_sparse, AMR_SUBFRAME_SIZE);
1045 ff_acelp_apply_order_2_transfer_function(buf_out, buf_out, highpass_zeros,
1047 highpass_gain * AMR_SAMPLE_SCALE,
1048 p->high_pass_mem, AMR_BLOCK_SIZE);
1050 /* Update averaged lsf vector (used for fixed gain smoothing).
1052 * Note that lsf_avg should not incorporate the current frame's LSFs
1053 * for fixed_gain_smooth.
1054 * The specification has an incorrect formula: the reference decoder uses
1055 * qbar(n-1) rather than qbar(n) in section 6.1(4) equation 71. */
1056 ff_weighted_vector_sumf(p->lsf_avg, p->lsf_avg, p->lsf_q[3],
1057 0.84, 0.16, LP_FILTER_ORDER);
1059 /* report how many samples we got */
1060 *data_size = AMR_BLOCK_SIZE * sizeof(float);
1062 /* return the amount of bytes consumed if everything was OK */
1063 return frame_sizes_nb[p->cur_frame_mode] + 1; // +7 for rounding and +8 for TOC
1067 AVCodec amrnb_decoder = {
1069 .type = AVMEDIA_TYPE_AUDIO,
1070 .id = CODEC_ID_AMR_NB,
1071 .priv_data_size = sizeof(AMRContext),
1072 .init = amrnb_decode_init,
1073 .decode = amrnb_decode_frame,
1074 .long_name = NULL_IF_CONFIG_SMALL("Adaptive Multi-Rate NarrowBand"),
1075 .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_FLT,SAMPLE_FMT_NONE},