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"
57 #include "amrnbdata.h"
59 #define AMR_BLOCK_SIZE 160 ///< samples per frame
60 #define AMR_SAMPLE_BOUND 32768.0 ///< threshold for synthesis overflow
63 * Scale from constructed speech to [-1,1]
65 * AMR is designed to produce 16-bit PCM samples (3GPP TS 26.090 4.2) but
66 * upscales by two (section 6.2.2).
68 * Fundamentally, this scale is determined by energy_mean through
69 * the fixed vector contribution to the excitation vector.
71 #define AMR_SAMPLE_SCALE (2.0 / 32768.0)
73 /** Prediction factor for 12.2kbit/s mode */
74 #define PRED_FAC_MODE_12k2 0.65
76 #define LSF_R_FAC (8000.0 / 32768.0) ///< LSF residual tables to Hertz
77 #define MIN_LSF_SPACING (50.0488 / 8000.0) ///< Ensures stability of LPC filter
78 #define PITCH_LAG_MIN_MODE_12k2 18 ///< Lower bound on decoded lag search in 12.2kbit/s mode
80 /** Initial energy in dB. Also used for bad frames (unimplemented). */
81 #define MIN_ENERGY -14.0
83 /** Maximum sharpening factor
85 * The specification says 0.8, which should be 13107, but the reference C code
86 * uses 13017 instead. (Amusingly the same applies to SHARP_MAX in g729dec.c.)
88 #define SHARP_MAX 0.79449462890625
90 /** Number of impulse response coefficients used for tilt factor */
91 #define AMR_TILT_RESPONSE 22
92 /** Tilt factor = 1st reflection coefficient * gamma_t */
93 #define AMR_TILT_GAMMA_T 0.8
94 /** Adaptive gain control factor used in post-filter */
95 #define AMR_AGC_ALPHA 0.9
97 typedef struct AMRContext {
98 AMRNBFrame frame; ///< decoded AMR parameters (lsf coefficients, codebook indexes, etc)
99 uint8_t bad_frame_indicator; ///< bad frame ? 1 : 0
100 enum Mode cur_frame_mode;
102 int16_t prev_lsf_r[LP_FILTER_ORDER]; ///< residual LSF vector from previous subframe
103 double lsp[4][LP_FILTER_ORDER]; ///< lsp vectors from current frame
104 double prev_lsp_sub4[LP_FILTER_ORDER]; ///< lsp vector for the 4th subframe of the previous frame
106 float lsf_q[4][LP_FILTER_ORDER]; ///< Interpolated LSF vector for fixed gain smoothing
107 float lsf_avg[LP_FILTER_ORDER]; ///< vector of averaged lsf vector
109 float lpc[4][LP_FILTER_ORDER]; ///< lpc coefficient vectors for 4 subframes
111 uint8_t pitch_lag_int; ///< integer part of pitch lag from current subframe
113 float excitation_buf[PITCH_DELAY_MAX + LP_FILTER_ORDER + 1 + AMR_SUBFRAME_SIZE]; ///< current excitation and all necessary excitation history
114 float *excitation; ///< pointer to the current excitation vector in excitation_buf
116 float pitch_vector[AMR_SUBFRAME_SIZE]; ///< adaptive code book (pitch) vector
117 float fixed_vector[AMR_SUBFRAME_SIZE]; ///< algebraic codebook (fixed) vector (must be kept zero between frames)
119 float prediction_error[4]; ///< quantified prediction errors {20log10(^gamma_gc)} for previous four subframes
120 float pitch_gain[5]; ///< quantified pitch gains for the current and previous four subframes
121 float fixed_gain[5]; ///< quantified fixed gains for the current and previous four subframes
123 float beta; ///< previous pitch_gain, bounded by [0.0,SHARP_MAX]
124 uint8_t diff_count; ///< the number of subframes for which diff has been above 0.65
125 uint8_t hang_count; ///< the number of subframes since a hangover period started
127 float prev_sparse_fixed_gain; ///< previous fixed gain; used by anti-sparseness processing to determine "onset"
128 uint8_t prev_ir_filter_nr; ///< previous impulse response filter "impNr": 0 - strong, 1 - medium, 2 - none
129 uint8_t ir_filter_onset; ///< flag for impulse response filter strength
131 float postfilter_mem[10]; ///< previous intermediate values in the formant filter
132 float tilt_mem; ///< previous input to tilt compensation filter
133 float postfilter_agc; ///< previous factor used for adaptive gain control
134 float high_pass_mem[2]; ///< previous intermediate values in the high-pass filter
136 float samples_in[LP_FILTER_ORDER + AMR_SUBFRAME_SIZE]; ///< floating point samples
140 /** Double version of ff_weighted_vector_sumf() */
141 static void weighted_vector_sumd(double *out, const double *in_a,
142 const double *in_b, double weight_coeff_a,
143 double weight_coeff_b, int length)
147 for (i = 0; i < length; i++)
148 out[i] = weight_coeff_a * in_a[i]
149 + weight_coeff_b * in_b[i];
152 static av_cold int amrnb_decode_init(AVCodecContext *avctx)
154 AMRContext *p = avctx->priv_data;
157 avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
159 // p->excitation always points to the same position in p->excitation_buf
160 p->excitation = &p->excitation_buf[PITCH_DELAY_MAX + LP_FILTER_ORDER + 1];
162 for (i = 0; i < LP_FILTER_ORDER; i++) {
163 p->prev_lsp_sub4[i] = lsp_sub4_init[i] * 1000 / (float)(1 << 15);
164 p->lsf_avg[i] = p->lsf_q[3][i] = lsp_avg_init[i] / (float)(1 << 15);
167 for (i = 0; i < 4; i++)
168 p->prediction_error[i] = MIN_ENERGY;
175 * Unpack an RFC4867 speech frame into the AMR frame mode and parameters.
177 * The order of speech bits is specified by 3GPP TS 26.101.
179 * @param p the context
180 * @param buf pointer to the input buffer
181 * @param buf_size size of the input buffer
183 * @return the frame mode
185 static enum Mode unpack_bitstream(AMRContext *p, const uint8_t *buf,
191 init_get_bits(&gb, buf, buf_size * 8);
193 // Decode the first octet.
194 skip_bits(&gb, 1); // padding bit
195 mode = get_bits(&gb, 4); // frame type
196 p->bad_frame_indicator = !get_bits1(&gb); // quality bit
197 skip_bits(&gb, 2); // two padding bits
200 ff_amr_bit_reorder((uint16_t *) &p->frame, sizeof(AMRNBFrame), buf + 1,
201 amr_unpacking_bitmaps_per_mode[mode]);
207 /// @defgroup amr_lpc_decoding AMR pitch LPC coefficient decoding functions
211 * Interpolate the LSF vector (used for fixed gain smoothing).
212 * The interpolation is done over all four subframes even in MODE_12k2.
214 * @param[in,out] lsf_q LSFs in [0,1] for each subframe
215 * @param[in] lsf_new New LSFs in [0,1] for subframe 4
217 static void interpolate_lsf(float lsf_q[4][LP_FILTER_ORDER], float *lsf_new)
221 for (i = 0; i < 4; i++)
222 ff_weighted_vector_sumf(lsf_q[i], lsf_q[3], lsf_new,
223 0.25 * (3 - i), 0.25 * (i + 1),
228 * Decode a set of 5 split-matrix quantized lsf indexes into an lsp vector.
230 * @param p the context
231 * @param lsp output LSP vector
232 * @param lsf_no_r LSF vector without the residual vector added
233 * @param lsf_quantizer pointers to LSF dictionary tables
234 * @param quantizer_offset offset in tables
235 * @param sign for the 3 dictionary table
236 * @param update store data for computing the next frame's LSFs
238 static void lsf2lsp_for_mode12k2(AMRContext *p, double lsp[LP_FILTER_ORDER],
239 const float lsf_no_r[LP_FILTER_ORDER],
240 const int16_t *lsf_quantizer[5],
241 const int quantizer_offset,
242 const int sign, const int update)
244 int16_t lsf_r[LP_FILTER_ORDER]; // residual LSF vector
245 float lsf_q[LP_FILTER_ORDER]; // quantified LSF vector
248 for (i = 0; i < LP_FILTER_ORDER >> 1; i++)
249 memcpy(&lsf_r[i << 1], &lsf_quantizer[i][quantizer_offset],
258 memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(*lsf_r));
260 for (i = 0; i < LP_FILTER_ORDER; i++)
261 lsf_q[i] = lsf_r[i] * (LSF_R_FAC / 8000.0) + lsf_no_r[i] * (1.0 / 8000.0);
263 ff_set_min_dist_lsf(lsf_q, MIN_LSF_SPACING, LP_FILTER_ORDER);
266 interpolate_lsf(p->lsf_q, lsf_q);
268 ff_acelp_lsf2lspd(lsp, lsf_q, LP_FILTER_ORDER);
272 * Decode a set of 5 split-matrix quantized lsf indexes into 2 lsp vectors.
274 * @param p pointer to the AMRContext
276 static void lsf2lsp_5(AMRContext *p)
278 const uint16_t *lsf_param = p->frame.lsf;
279 float lsf_no_r[LP_FILTER_ORDER]; // LSFs without the residual vector
280 const int16_t *lsf_quantizer[5];
283 lsf_quantizer[0] = lsf_5_1[lsf_param[0]];
284 lsf_quantizer[1] = lsf_5_2[lsf_param[1]];
285 lsf_quantizer[2] = lsf_5_3[lsf_param[2] >> 1];
286 lsf_quantizer[3] = lsf_5_4[lsf_param[3]];
287 lsf_quantizer[4] = lsf_5_5[lsf_param[4]];
289 for (i = 0; i < LP_FILTER_ORDER; i++)
290 lsf_no_r[i] = p->prev_lsf_r[i] * LSF_R_FAC * PRED_FAC_MODE_12k2 + lsf_5_mean[i];
292 lsf2lsp_for_mode12k2(p, p->lsp[1], lsf_no_r, lsf_quantizer, 0, lsf_param[2] & 1, 0);
293 lsf2lsp_for_mode12k2(p, p->lsp[3], lsf_no_r, lsf_quantizer, 2, lsf_param[2] & 1, 1);
295 // interpolate LSP vectors at subframes 1 and 3
296 weighted_vector_sumd(p->lsp[0], p->prev_lsp_sub4, p->lsp[1], 0.5, 0.5, LP_FILTER_ORDER);
297 weighted_vector_sumd(p->lsp[2], p->lsp[1] , p->lsp[3], 0.5, 0.5, LP_FILTER_ORDER);
301 * Decode a set of 3 split-matrix quantized lsf indexes into an lsp vector.
303 * @param p pointer to the AMRContext
305 static void lsf2lsp_3(AMRContext *p)
307 const uint16_t *lsf_param = p->frame.lsf;
308 int16_t lsf_r[LP_FILTER_ORDER]; // residual LSF vector
309 float lsf_q[LP_FILTER_ORDER]; // quantified LSF vector
310 const int16_t *lsf_quantizer;
313 lsf_quantizer = (p->cur_frame_mode == MODE_7k95 ? lsf_3_1_MODE_7k95 : lsf_3_1)[lsf_param[0]];
314 memcpy(lsf_r, lsf_quantizer, 3 * sizeof(*lsf_r));
316 lsf_quantizer = lsf_3_2[lsf_param[1] << (p->cur_frame_mode <= MODE_5k15)];
317 memcpy(lsf_r + 3, lsf_quantizer, 3 * sizeof(*lsf_r));
319 lsf_quantizer = (p->cur_frame_mode <= MODE_5k15 ? lsf_3_3_MODE_5k15 : lsf_3_3)[lsf_param[2]];
320 memcpy(lsf_r + 6, lsf_quantizer, 4 * sizeof(*lsf_r));
322 // calculate mean-removed LSF vector and add mean
323 for (i = 0; i < LP_FILTER_ORDER; i++)
324 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);
326 ff_set_min_dist_lsf(lsf_q, MIN_LSF_SPACING, LP_FILTER_ORDER);
328 // store data for computing the next frame's LSFs
329 interpolate_lsf(p->lsf_q, lsf_q);
330 memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(*lsf_r));
332 ff_acelp_lsf2lspd(p->lsp[3], lsf_q, LP_FILTER_ORDER);
334 // interpolate LSP vectors at subframes 1, 2 and 3
335 for (i = 1; i <= 3; i++)
336 for(j = 0; j < LP_FILTER_ORDER; j++)
337 p->lsp[i-1][j] = p->prev_lsp_sub4[j] +
338 (p->lsp[3][j] - p->prev_lsp_sub4[j]) * 0.25 * i;
344 /// @defgroup amr_pitch_vector_decoding AMR pitch vector decoding functions
348 * Like ff_decode_pitch_lag(), but with 1/6 resolution
350 static void decode_pitch_lag_1_6(int *lag_int, int *lag_frac, int pitch_index,
351 const int prev_lag_int, const int subframe)
353 if (subframe == 0 || subframe == 2) {
354 if (pitch_index < 463) {
355 *lag_int = (pitch_index + 107) * 10923 >> 16;
356 *lag_frac = pitch_index - *lag_int * 6 + 105;
358 *lag_int = pitch_index - 368;
362 *lag_int = ((pitch_index + 5) * 10923 >> 16) - 1;
363 *lag_frac = pitch_index - *lag_int * 6 - 3;
364 *lag_int += av_clip(prev_lag_int - 5, PITCH_LAG_MIN_MODE_12k2,
365 PITCH_DELAY_MAX - 9);
369 static void decode_pitch_vector(AMRContext *p,
370 const AMRNBSubframe *amr_subframe,
373 int pitch_lag_int, pitch_lag_frac;
374 enum Mode mode = p->cur_frame_mode;
376 if (p->cur_frame_mode == MODE_12k2) {
377 decode_pitch_lag_1_6(&pitch_lag_int, &pitch_lag_frac,
378 amr_subframe->p_lag, p->pitch_lag_int,
381 ff_decode_pitch_lag(&pitch_lag_int, &pitch_lag_frac,
383 p->pitch_lag_int, subframe,
384 mode != MODE_4k75 && mode != MODE_5k15,
385 mode <= MODE_6k7 ? 4 : (mode == MODE_7k95 ? 5 : 6));
387 p->pitch_lag_int = pitch_lag_int; // store previous lag in a uint8_t
389 pitch_lag_frac <<= (p->cur_frame_mode != MODE_12k2);
391 pitch_lag_int += pitch_lag_frac > 0;
393 /* Calculate the pitch vector by interpolating the past excitation at the
394 pitch lag using a b60 hamming windowed sinc function. */
395 ff_acelp_interpolatef(p->excitation, p->excitation + 1 - pitch_lag_int,
397 pitch_lag_frac + 6 - 6*(pitch_lag_frac > 0),
398 10, AMR_SUBFRAME_SIZE);
400 memcpy(p->pitch_vector, p->excitation, AMR_SUBFRAME_SIZE * sizeof(float));
406 /// @defgroup amr_algebraic_code_book AMR algebraic code book (fixed) vector decoding functions
410 * Decode a 10-bit algebraic codebook index from a 10.2 kbit/s frame.
412 static void decode_10bit_pulse(int code, int pulse_position[8],
413 int i1, int i2, int i3)
415 // coded using 7+3 bits with the 3 LSBs being, individually, the LSB of 1 of
416 // the 3 pulses and the upper 7 bits being coded in base 5
417 const uint8_t *positions = base_five_table[code >> 3];
418 pulse_position[i1] = (positions[2] << 1) + ( code & 1);
419 pulse_position[i2] = (positions[1] << 1) + ((code >> 1) & 1);
420 pulse_position[i3] = (positions[0] << 1) + ((code >> 2) & 1);
424 * Decode the algebraic codebook index to pulse positions and signs and
425 * construct the algebraic codebook vector for MODE_10k2.
427 * @param fixed_index positions of the eight pulses
428 * @param fixed_sparse pointer to the algebraic codebook vector
430 static void decode_8_pulses_31bits(const int16_t *fixed_index,
431 AMRFixed *fixed_sparse)
433 int pulse_position[8];
436 decode_10bit_pulse(fixed_index[4], pulse_position, 0, 4, 1);
437 decode_10bit_pulse(fixed_index[5], pulse_position, 2, 6, 5);
439 // coded using 5+2 bits with the 2 LSBs being, individually, the LSB of 1 of
440 // the 2 pulses and the upper 5 bits being coded in base 5
441 temp = ((fixed_index[6] >> 2) * 25 + 12) >> 5;
442 pulse_position[3] = temp % 5;
443 pulse_position[7] = temp / 5;
444 if (pulse_position[7] & 1)
445 pulse_position[3] = 4 - pulse_position[3];
446 pulse_position[3] = (pulse_position[3] << 1) + ( fixed_index[6] & 1);
447 pulse_position[7] = (pulse_position[7] << 1) + ((fixed_index[6] >> 1) & 1);
450 for (i = 0; i < 4; i++) {
451 const int pos1 = (pulse_position[i] << 2) + i;
452 const int pos2 = (pulse_position[i + 4] << 2) + i;
453 const float sign = fixed_index[i] ? -1.0 : 1.0;
454 fixed_sparse->x[i ] = pos1;
455 fixed_sparse->x[i + 4] = pos2;
456 fixed_sparse->y[i ] = sign;
457 fixed_sparse->y[i + 4] = pos2 < pos1 ? -sign : sign;
462 * Decode the algebraic codebook index to pulse positions and signs,
463 * then construct the algebraic codebook vector.
465 * nb of pulses | bits encoding pulses
466 * For MODE_4k75 or MODE_5k15, 2 | 1-3, 4-6, 7
467 * MODE_5k9, 2 | 1, 2-4, 5-6, 7-9
468 * MODE_6k7, 3 | 1-3, 4, 5-7, 8, 9-11
469 * MODE_7k4 or MODE_7k95, 4 | 1-3, 4-6, 7-9, 10, 11-13
471 * @param fixed_sparse pointer to the algebraic codebook vector
472 * @param pulses algebraic codebook indexes
473 * @param mode mode of the current frame
474 * @param subframe current subframe number
476 static void decode_fixed_sparse(AMRFixed *fixed_sparse, const uint16_t *pulses,
477 const enum Mode mode, const int subframe)
479 assert(MODE_4k75 <= mode && mode <= MODE_12k2);
481 if (mode == MODE_12k2) {
482 ff_decode_10_pulses_35bits(pulses, fixed_sparse, gray_decode, 5, 3);
483 } else if (mode == MODE_10k2) {
484 decode_8_pulses_31bits(pulses, fixed_sparse);
486 int *pulse_position = fixed_sparse->x;
488 const int fixed_index = pulses[0];
490 if (mode <= MODE_5k15) {
491 pulse_subset = ((fixed_index >> 3) & 8) + (subframe << 1);
492 pulse_position[0] = ( fixed_index & 7) * 5 + track_position[pulse_subset];
493 pulse_position[1] = ((fixed_index >> 3) & 7) * 5 + track_position[pulse_subset + 1];
495 } else if (mode == MODE_5k9) {
496 pulse_subset = ((fixed_index & 1) << 1) + 1;
497 pulse_position[0] = ((fixed_index >> 1) & 7) * 5 + pulse_subset;
498 pulse_subset = (fixed_index >> 4) & 3;
499 pulse_position[1] = ((fixed_index >> 6) & 7) * 5 + pulse_subset + (pulse_subset == 3 ? 1 : 0);
500 fixed_sparse->n = pulse_position[0] == pulse_position[1] ? 1 : 2;
501 } else if (mode == MODE_6k7) {
502 pulse_position[0] = (fixed_index & 7) * 5;
503 pulse_subset = (fixed_index >> 2) & 2;
504 pulse_position[1] = ((fixed_index >> 4) & 7) * 5 + pulse_subset + 1;
505 pulse_subset = (fixed_index >> 6) & 2;
506 pulse_position[2] = ((fixed_index >> 8) & 7) * 5 + pulse_subset + 2;
508 } else { // mode <= MODE_7k95
509 pulse_position[0] = gray_decode[ fixed_index & 7];
510 pulse_position[1] = gray_decode[(fixed_index >> 3) & 7] + 1;
511 pulse_position[2] = gray_decode[(fixed_index >> 6) & 7] + 2;
512 pulse_subset = (fixed_index >> 9) & 1;
513 pulse_position[3] = gray_decode[(fixed_index >> 10) & 7] + pulse_subset + 3;
516 for (i = 0; i < fixed_sparse->n; i++)
517 fixed_sparse->y[i] = (pulses[1] >> i) & 1 ? 1.0 : -1.0;
522 * Apply pitch lag to obtain the sharpened fixed vector (section 6.1.2)
524 * @param p the context
525 * @param subframe unpacked amr subframe
526 * @param mode mode of the current frame
527 * @param fixed_sparse sparse respresentation of the fixed vector
529 static void pitch_sharpening(AMRContext *p, int subframe, enum Mode mode,
530 AMRFixed *fixed_sparse)
532 // The spec suggests the current pitch gain is always used, but in other
533 // modes the pitch and codebook gains are joinly quantized (sec 5.8.2)
534 // so the codebook gain cannot depend on the quantized pitch gain.
535 if (mode == MODE_12k2)
536 p->beta = FFMIN(p->pitch_gain[4], 1.0);
538 fixed_sparse->pitch_lag = p->pitch_lag_int;
539 fixed_sparse->pitch_fac = p->beta;
541 // Save pitch sharpening factor for the next subframe
542 // MODE_4k75 only updates on the 2nd and 4th subframes - this follows from
543 // the fact that the gains for two subframes are jointly quantized.
544 if (mode != MODE_4k75 || subframe & 1)
545 p->beta = av_clipf(p->pitch_gain[4], 0.0, SHARP_MAX);
550 /// @defgroup amr_gain_decoding AMR gain decoding functions
554 * fixed gain smoothing
555 * Note that where the spec specifies the "spectrum in the q domain"
556 * in section 6.1.4, in fact frequencies should be used.
558 * @param p the context
559 * @param lsf LSFs for the current subframe, in the range [0,1]
560 * @param lsf_avg averaged LSFs
561 * @param mode mode of the current frame
563 * @return fixed gain smoothed
565 static float fixed_gain_smooth(AMRContext *p , const float *lsf,
566 const float *lsf_avg, const enum Mode mode)
571 for (i = 0; i < LP_FILTER_ORDER; i++)
572 diff += fabs(lsf_avg[i] - lsf[i]) / lsf_avg[i];
574 // If diff is large for ten subframes, disable smoothing for a 40-subframe
580 if (p->diff_count > 10) {
582 p->diff_count--; // don't let diff_count overflow
585 if (p->hang_count < 40) {
587 } else if (mode < MODE_7k4 || mode == MODE_10k2) {
588 const float smoothing_factor = av_clipf(4.0 * diff - 1.6, 0.0, 1.0);
589 const float fixed_gain_mean = (p->fixed_gain[0] + p->fixed_gain[1] +
590 p->fixed_gain[2] + p->fixed_gain[3] +
591 p->fixed_gain[4]) * 0.2;
592 return smoothing_factor * p->fixed_gain[4] +
593 (1.0 - smoothing_factor) * fixed_gain_mean;
595 return p->fixed_gain[4];
599 * Decode pitch gain and fixed gain factor (part of section 6.1.3).
601 * @param p the context
602 * @param amr_subframe unpacked amr subframe
603 * @param mode mode of the current frame
604 * @param subframe current subframe number
605 * @param fixed_gain_factor decoded gain correction factor
607 static void decode_gains(AMRContext *p, const AMRNBSubframe *amr_subframe,
608 const enum Mode mode, const int subframe,
609 float *fixed_gain_factor)
611 if (mode == MODE_12k2 || mode == MODE_7k95) {
612 p->pitch_gain[4] = qua_gain_pit [amr_subframe->p_gain ]
614 *fixed_gain_factor = qua_gain_code[amr_subframe->fixed_gain]
617 const uint16_t *gains;
619 if (mode >= MODE_6k7) {
620 gains = gains_high[amr_subframe->p_gain];
621 } else if (mode >= MODE_5k15) {
622 gains = gains_low [amr_subframe->p_gain];
624 // gain index is only coded in subframes 0,2 for MODE_4k75
625 gains = gains_MODE_4k75[(p->frame.subframe[subframe & 2].p_gain << 1) + (subframe & 1)];
628 p->pitch_gain[4] = gains[0] * (1.0 / 16384.0);
629 *fixed_gain_factor = gains[1] * (1.0 / 4096.0);
636 /// @defgroup amr_pre_processing AMR pre-processing functions
640 * Circularly convolve a sparse fixed vector with a phase dispersion impulse
641 * response filter (D.6.2 of G.729 and 6.1.5 of AMR).
643 * @param out vector with filter applied
644 * @param in source vector
645 * @param filter phase filter coefficients
647 * out[n] = sum(i,0,len-1){ in[i] * filter[(len + n - i)%len] }
649 static void apply_ir_filter(float *out, const AMRFixed *in,
652 float filter1[AMR_SUBFRAME_SIZE], //!< filters at pitch lag*1 and *2
653 filter2[AMR_SUBFRAME_SIZE];
654 int lag = in->pitch_lag;
655 float fac = in->pitch_fac;
658 if (lag < AMR_SUBFRAME_SIZE) {
659 ff_celp_circ_addf(filter1, filter, filter, lag, fac,
662 if (lag < AMR_SUBFRAME_SIZE >> 1)
663 ff_celp_circ_addf(filter2, filter, filter1, lag, fac,
667 memset(out, 0, sizeof(float) * AMR_SUBFRAME_SIZE);
668 for (i = 0; i < in->n; i++) {
671 const float *filterp;
673 if (x >= AMR_SUBFRAME_SIZE - lag) {
675 } else if (x >= AMR_SUBFRAME_SIZE - (lag << 1)) {
680 ff_celp_circ_addf(out, out, filterp, x, y, AMR_SUBFRAME_SIZE);
685 * Reduce fixed vector sparseness by smoothing with one of three IR filters.
686 * Also know as "adaptive phase dispersion".
688 * This implements 3GPP TS 26.090 section 6.1(5).
690 * @param p the context
691 * @param fixed_sparse algebraic codebook vector
692 * @param fixed_vector unfiltered fixed vector
693 * @param fixed_gain smoothed gain
694 * @param out space for modified vector if necessary
696 static const float *anti_sparseness(AMRContext *p, AMRFixed *fixed_sparse,
697 const float *fixed_vector,
698 float fixed_gain, float *out)
702 if (p->pitch_gain[4] < 0.6) {
703 ir_filter_nr = 0; // strong filtering
704 } else if (p->pitch_gain[4] < 0.9) {
705 ir_filter_nr = 1; // medium filtering
707 ir_filter_nr = 2; // no filtering
710 if (fixed_gain > 2.0 * p->prev_sparse_fixed_gain) {
711 p->ir_filter_onset = 2;
712 } else if (p->ir_filter_onset)
713 p->ir_filter_onset--;
715 if (!p->ir_filter_onset) {
718 for (i = 0; i < 5; i++)
719 if (p->pitch_gain[i] < 0.6)
724 if (ir_filter_nr > p->prev_ir_filter_nr + 1)
726 } else if (ir_filter_nr < 2)
729 // Disable filtering for very low level of fixed_gain.
730 // Note this step is not specified in the technical description but is in
731 // the reference source in the function Ph_disp.
732 if (fixed_gain < 5.0)
735 if (p->cur_frame_mode != MODE_7k4 && p->cur_frame_mode < MODE_10k2
736 && ir_filter_nr < 2) {
737 apply_ir_filter(out, fixed_sparse,
738 (p->cur_frame_mode == MODE_7k95 ?
739 ir_filters_lookup_MODE_7k95 :
740 ir_filters_lookup)[ir_filter_nr]);
744 // update ir filter strength history
745 p->prev_ir_filter_nr = ir_filter_nr;
746 p->prev_sparse_fixed_gain = fixed_gain;
754 /// @defgroup amr_synthesis AMR synthesis functions
758 * Conduct 10th order linear predictive coding synthesis.
760 * @param p pointer to the AMRContext
761 * @param lpc pointer to the LPC coefficients
762 * @param fixed_gain fixed codebook gain for synthesis
763 * @param fixed_vector algebraic codebook vector
764 * @param samples pointer to the output speech samples
765 * @param overflow 16-bit overflow flag
767 static int synthesis(AMRContext *p, float *lpc,
768 float fixed_gain, const float *fixed_vector,
769 float *samples, uint8_t overflow)
772 float excitation[AMR_SUBFRAME_SIZE];
774 // if an overflow has been detected, the pitch vector is scaled down by a
777 for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
778 p->pitch_vector[i] *= 0.25;
780 ff_weighted_vector_sumf(excitation, p->pitch_vector, fixed_vector,
781 p->pitch_gain[4], fixed_gain, AMR_SUBFRAME_SIZE);
783 // emphasize pitch vector contribution
784 if (p->pitch_gain[4] > 0.5 && !overflow) {
785 float energy = ff_dot_productf(excitation, excitation,
789 (p->cur_frame_mode == MODE_12k2 ?
790 0.25 * FFMIN(p->pitch_gain[4], 1.0) :
791 0.5 * FFMIN(p->pitch_gain[4], SHARP_MAX));
793 for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
794 excitation[i] += pitch_factor * p->pitch_vector[i];
796 ff_scale_vector_to_given_sum_of_squares(excitation, excitation, energy,
800 ff_celp_lp_synthesis_filterf(samples, lpc, excitation, AMR_SUBFRAME_SIZE,
804 for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
805 if (fabsf(samples[i]) > AMR_SAMPLE_BOUND) {
815 /// @defgroup amr_update AMR update functions
819 * Update buffers and history at the end of decoding a subframe.
821 * @param p pointer to the AMRContext
823 static void update_state(AMRContext *p)
825 memcpy(p->prev_lsp_sub4, p->lsp[3], LP_FILTER_ORDER * sizeof(p->lsp[3][0]));
827 memmove(&p->excitation_buf[0], &p->excitation_buf[AMR_SUBFRAME_SIZE],
828 (PITCH_DELAY_MAX + LP_FILTER_ORDER + 1) * sizeof(float));
830 memmove(&p->pitch_gain[0], &p->pitch_gain[1], 4 * sizeof(float));
831 memmove(&p->fixed_gain[0], &p->fixed_gain[1], 4 * sizeof(float));
833 memmove(&p->samples_in[0], &p->samples_in[AMR_SUBFRAME_SIZE],
834 LP_FILTER_ORDER * sizeof(float));
840 /// @defgroup amr_postproc AMR Post processing functions
844 * Get the tilt factor of a formant filter from its transfer function
846 * @param lpc_n LP_FILTER_ORDER coefficients of the numerator
847 * @param lpc_d LP_FILTER_ORDER coefficients of the denominator
849 static float tilt_factor(float *lpc_n, float *lpc_d)
851 float rh0, rh1; // autocorrelation at lag 0 and 1
853 // LP_FILTER_ORDER prior zeros are needed for ff_celp_lp_synthesis_filterf
854 float impulse_buffer[LP_FILTER_ORDER + AMR_TILT_RESPONSE] = { 0 };
855 float *hf = impulse_buffer + LP_FILTER_ORDER; // start of impulse response
858 memcpy(hf + 1, lpc_n, sizeof(float) * LP_FILTER_ORDER);
859 ff_celp_lp_synthesis_filterf(hf, lpc_d, hf, AMR_TILT_RESPONSE,
862 rh0 = ff_dot_productf(hf, hf, AMR_TILT_RESPONSE);
863 rh1 = ff_dot_productf(hf, hf + 1, AMR_TILT_RESPONSE - 1);
865 // The spec only specifies this check for 12.2 and 10.2 kbit/s
866 // modes. But in the ref source the tilt is always non-negative.
867 return rh1 >= 0.0 ? rh1 / rh0 * AMR_TILT_GAMMA_T : 0.0;
871 * Perform adaptive post-filtering to enhance the quality of the speech.
874 * @param p pointer to the AMRContext
875 * @param lpc interpolated LP coefficients for this subframe
876 * @param buf_out output of the filter
878 static void postfilter(AMRContext *p, float *lpc, float *buf_out)
881 float *samples = p->samples_in + LP_FILTER_ORDER; // Start of input
883 float speech_gain = ff_dot_productf(samples, samples,
886 float pole_out[AMR_SUBFRAME_SIZE + LP_FILTER_ORDER]; // Output of pole filter
887 const float *gamma_n, *gamma_d; // Formant filter factor table
888 float lpc_n[LP_FILTER_ORDER], lpc_d[LP_FILTER_ORDER]; // Transfer function coefficients
890 if (p->cur_frame_mode == MODE_12k2 || p->cur_frame_mode == MODE_10k2) {
891 gamma_n = ff_pow_0_7;
892 gamma_d = ff_pow_0_75;
894 gamma_n = ff_pow_0_55;
895 gamma_d = ff_pow_0_7;
898 for (i = 0; i < LP_FILTER_ORDER; i++) {
899 lpc_n[i] = lpc[i] * gamma_n[i];
900 lpc_d[i] = lpc[i] * gamma_d[i];
903 memcpy(pole_out, p->postfilter_mem, sizeof(float) * LP_FILTER_ORDER);
904 ff_celp_lp_synthesis_filterf(pole_out + LP_FILTER_ORDER, lpc_d, samples,
905 AMR_SUBFRAME_SIZE, LP_FILTER_ORDER);
906 memcpy(p->postfilter_mem, pole_out + AMR_SUBFRAME_SIZE,
907 sizeof(float) * LP_FILTER_ORDER);
909 ff_celp_lp_zero_synthesis_filterf(buf_out, lpc_n,
910 pole_out + LP_FILTER_ORDER,
911 AMR_SUBFRAME_SIZE, LP_FILTER_ORDER);
913 ff_tilt_compensation(&p->tilt_mem, tilt_factor(lpc_n, lpc_d), buf_out,
916 ff_adaptive_gain_control(buf_out, buf_out, speech_gain, AMR_SUBFRAME_SIZE,
917 AMR_AGC_ALPHA, &p->postfilter_agc);
922 static int amrnb_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
926 AMRContext *p = avctx->priv_data; // pointer to private data
927 const uint8_t *buf = avpkt->data;
928 int buf_size = avpkt->size;
929 float *buf_out = data; // pointer to the output data buffer
931 float fixed_gain_factor;
932 AMRFixed fixed_sparse = {0}; // fixed vector up to anti-sparseness processing
933 float spare_vector[AMR_SUBFRAME_SIZE]; // extra stack space to hold result from anti-sparseness processing
934 float synth_fixed_gain; // the fixed gain that synthesis should use
935 const float *synth_fixed_vector; // pointer to the fixed vector that synthesis should use
937 p->cur_frame_mode = unpack_bitstream(p, buf, buf_size);
938 if (p->cur_frame_mode == MODE_DTX) {
939 av_log_missing_feature(avctx, "dtx mode", 1);
943 if (p->cur_frame_mode == MODE_12k2) {
948 for (i = 0; i < 4; i++)
949 ff_acelp_lspd2lpc(p->lsp[i], p->lpc[i], 5);
951 for (subframe = 0; subframe < 4; subframe++) {
952 const AMRNBSubframe *amr_subframe = &p->frame.subframe[subframe];
954 decode_pitch_vector(p, amr_subframe, subframe);
956 decode_fixed_sparse(&fixed_sparse, amr_subframe->pulses,
957 p->cur_frame_mode, subframe);
959 // The fixed gain (section 6.1.3) depends on the fixed vector
960 // (section 6.1.2), but the fixed vector calculation uses
961 // pitch sharpening based on the on the pitch gain (section 6.1.3).
962 // So the correct order is: pitch gain, pitch sharpening, fixed gain.
963 decode_gains(p, amr_subframe, p->cur_frame_mode, subframe,
966 pitch_sharpening(p, subframe, p->cur_frame_mode, &fixed_sparse);
968 ff_set_fixed_vector(p->fixed_vector, &fixed_sparse, 1.0,
972 ff_amr_set_fixed_gain(fixed_gain_factor,
973 ff_dot_productf(p->fixed_vector, p->fixed_vector,
974 AMR_SUBFRAME_SIZE)/AMR_SUBFRAME_SIZE,
976 energy_mean[p->cur_frame_mode], energy_pred_fac);
978 // The excitation feedback is calculated without any processing such
979 // as fixed gain smoothing. This isn't mentioned in the specification.
980 for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
981 p->excitation[i] *= p->pitch_gain[4];
982 ff_set_fixed_vector(p->excitation, &fixed_sparse, p->fixed_gain[4],
985 // In the ref decoder, excitation is stored with no fractional bits.
986 // This step prevents buzz in silent periods. The ref encoder can
987 // emit long sequences with pitch factor greater than one. This
988 // creates unwanted feedback if the excitation vector is nonzero.
989 // (e.g. test sequence T19_795.COD in 3GPP TS 26.074)
990 for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
991 p->excitation[i] = truncf(p->excitation[i]);
993 // Smooth fixed gain.
994 // The specification is ambiguous, but in the reference source, the
995 // smoothed value is NOT fed back into later fixed gain smoothing.
996 synth_fixed_gain = fixed_gain_smooth(p, p->lsf_q[subframe],
997 p->lsf_avg, p->cur_frame_mode);
999 synth_fixed_vector = anti_sparseness(p, &fixed_sparse, p->fixed_vector,
1000 synth_fixed_gain, spare_vector);
1002 if (synthesis(p, p->lpc[subframe], synth_fixed_gain,
1003 synth_fixed_vector, &p->samples_in[LP_FILTER_ORDER], 0))
1004 // overflow detected -> rerun synthesis scaling pitch vector down
1005 // by a factor of 4, skipping pitch vector contribution emphasis
1006 // and adaptive gain control
1007 synthesis(p, p->lpc[subframe], synth_fixed_gain,
1008 synth_fixed_vector, &p->samples_in[LP_FILTER_ORDER], 1);
1010 postfilter(p, p->lpc[subframe], buf_out + subframe * AMR_SUBFRAME_SIZE);
1012 // update buffers and history
1013 ff_clear_fixed_vector(p->fixed_vector, &fixed_sparse, AMR_SUBFRAME_SIZE);
1017 ff_acelp_apply_order_2_transfer_function(buf_out, buf_out, highpass_zeros,
1019 highpass_gain * AMR_SAMPLE_SCALE,
1020 p->high_pass_mem, AMR_BLOCK_SIZE);
1022 /* Update averaged lsf vector (used for fixed gain smoothing).
1024 * Note that lsf_avg should not incorporate the current frame's LSFs
1025 * for fixed_gain_smooth.
1026 * The specification has an incorrect formula: the reference decoder uses
1027 * qbar(n-1) rather than qbar(n) in section 6.1(4) equation 71. */
1028 ff_weighted_vector_sumf(p->lsf_avg, p->lsf_avg, p->lsf_q[3],
1029 0.84, 0.16, LP_FILTER_ORDER);
1031 /* report how many samples we got */
1032 *data_size = AMR_BLOCK_SIZE * sizeof(float);
1034 /* return the amount of bytes consumed if everything was OK */
1035 return frame_sizes_nb[p->cur_frame_mode] + 1; // +7 for rounding and +8 for TOC
1039 AVCodec ff_amrnb_decoder = {
1041 .type = AVMEDIA_TYPE_AUDIO,
1042 .id = CODEC_ID_AMR_NB,
1043 .priv_data_size = sizeof(AMRContext),
1044 .init = amrnb_decode_init,
1045 .decode = amrnb_decode_frame,
1046 .long_name = NULL_IF_CONFIG_SMALL("Adaptive Multi-Rate NarrowBand"),
1047 .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE},