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 AVFrame avframe; ///< AVFrame for decoded samples
99 AMRNBFrame frame; ///< decoded AMR parameters (lsf coefficients, codebook indexes, etc)
100 uint8_t bad_frame_indicator; ///< bad frame ? 1 : 0
101 enum Mode cur_frame_mode;
103 int16_t prev_lsf_r[LP_FILTER_ORDER]; ///< residual LSF vector from previous subframe
104 double lsp[4][LP_FILTER_ORDER]; ///< lsp vectors from current frame
105 double prev_lsp_sub4[LP_FILTER_ORDER]; ///< lsp vector for the 4th subframe of the previous frame
107 float lsf_q[4][LP_FILTER_ORDER]; ///< Interpolated LSF vector for fixed gain smoothing
108 float lsf_avg[LP_FILTER_ORDER]; ///< vector of averaged lsf vector
110 float lpc[4][LP_FILTER_ORDER]; ///< lpc coefficient vectors for 4 subframes
112 uint8_t pitch_lag_int; ///< integer part of pitch lag from current subframe
114 float excitation_buf[PITCH_DELAY_MAX + LP_FILTER_ORDER + 1 + AMR_SUBFRAME_SIZE]; ///< current excitation and all necessary excitation history
115 float *excitation; ///< pointer to the current excitation vector in excitation_buf
117 float pitch_vector[AMR_SUBFRAME_SIZE]; ///< adaptive code book (pitch) vector
118 float fixed_vector[AMR_SUBFRAME_SIZE]; ///< algebraic codebook (fixed) vector (must be kept zero between frames)
120 float prediction_error[4]; ///< quantified prediction errors {20log10(^gamma_gc)} for previous four subframes
121 float pitch_gain[5]; ///< quantified pitch gains for the current and previous four subframes
122 float fixed_gain[5]; ///< quantified fixed gains for the current and previous four subframes
124 float beta; ///< previous pitch_gain, bounded by [0.0,SHARP_MAX]
125 uint8_t diff_count; ///< the number of subframes for which diff has been above 0.65
126 uint8_t hang_count; ///< the number of subframes since a hangover period started
128 float prev_sparse_fixed_gain; ///< previous fixed gain; used by anti-sparseness processing to determine "onset"
129 uint8_t prev_ir_filter_nr; ///< previous impulse response filter "impNr": 0 - strong, 1 - medium, 2 - none
130 uint8_t ir_filter_onset; ///< flag for impulse response filter strength
132 float postfilter_mem[10]; ///< previous intermediate values in the formant filter
133 float tilt_mem; ///< previous input to tilt compensation filter
134 float postfilter_agc; ///< previous factor used for adaptive gain control
135 float high_pass_mem[2]; ///< previous intermediate values in the high-pass filter
137 float samples_in[LP_FILTER_ORDER + AMR_SUBFRAME_SIZE]; ///< floating point samples
141 /** Double version of ff_weighted_vector_sumf() */
142 static void weighted_vector_sumd(double *out, const double *in_a,
143 const double *in_b, double weight_coeff_a,
144 double weight_coeff_b, int length)
148 for (i = 0; i < length; i++)
149 out[i] = weight_coeff_a * in_a[i]
150 + weight_coeff_b * in_b[i];
153 static av_cold int amrnb_decode_init(AVCodecContext *avctx)
155 AMRContext *p = avctx->priv_data;
158 avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
160 // p->excitation always points to the same position in p->excitation_buf
161 p->excitation = &p->excitation_buf[PITCH_DELAY_MAX + LP_FILTER_ORDER + 1];
163 for (i = 0; i < LP_FILTER_ORDER; i++) {
164 p->prev_lsp_sub4[i] = lsp_sub4_init[i] * 1000 / (float)(1 << 15);
165 p->lsf_avg[i] = p->lsf_q[3][i] = lsp_avg_init[i] / (float)(1 << 15);
168 for (i = 0; i < 4; i++)
169 p->prediction_error[i] = MIN_ENERGY;
171 avcodec_get_frame_defaults(&p->avframe);
172 avctx->coded_frame = &p->avframe;
179 * Unpack an RFC4867 speech frame into the AMR frame mode and parameters.
181 * The order of speech bits is specified by 3GPP TS 26.101.
183 * @param p the context
184 * @param buf pointer to the input buffer
185 * @param buf_size size of the input buffer
187 * @return the frame mode
189 static enum Mode unpack_bitstream(AMRContext *p, const uint8_t *buf,
195 init_get_bits(&gb, buf, buf_size * 8);
197 // Decode the first octet.
198 skip_bits(&gb, 1); // padding bit
199 mode = get_bits(&gb, 4); // frame type
200 p->bad_frame_indicator = !get_bits1(&gb); // quality bit
201 skip_bits(&gb, 2); // two padding bits
204 ff_amr_bit_reorder((uint16_t *) &p->frame, sizeof(AMRNBFrame), buf + 1,
205 amr_unpacking_bitmaps_per_mode[mode]);
211 /// @name AMR pitch LPC coefficient decoding functions
215 * Interpolate the LSF vector (used for fixed gain smoothing).
216 * The interpolation is done over all four subframes even in MODE_12k2.
218 * @param[in,out] lsf_q LSFs in [0,1] for each subframe
219 * @param[in] lsf_new New LSFs in [0,1] for subframe 4
221 static void interpolate_lsf(float lsf_q[4][LP_FILTER_ORDER], float *lsf_new)
225 for (i = 0; i < 4; i++)
226 ff_weighted_vector_sumf(lsf_q[i], lsf_q[3], lsf_new,
227 0.25 * (3 - i), 0.25 * (i + 1),
232 * Decode a set of 5 split-matrix quantized lsf indexes into an lsp vector.
234 * @param p the context
235 * @param lsp output LSP vector
236 * @param lsf_no_r LSF vector without the residual vector added
237 * @param lsf_quantizer pointers to LSF dictionary tables
238 * @param quantizer_offset offset in tables
239 * @param sign for the 3 dictionary table
240 * @param update store data for computing the next frame's LSFs
242 static void lsf2lsp_for_mode12k2(AMRContext *p, double lsp[LP_FILTER_ORDER],
243 const float lsf_no_r[LP_FILTER_ORDER],
244 const int16_t *lsf_quantizer[5],
245 const int quantizer_offset,
246 const int sign, const int update)
248 int16_t lsf_r[LP_FILTER_ORDER]; // residual LSF vector
249 float lsf_q[LP_FILTER_ORDER]; // quantified LSF vector
252 for (i = 0; i < LP_FILTER_ORDER >> 1; i++)
253 memcpy(&lsf_r[i << 1], &lsf_quantizer[i][quantizer_offset],
262 memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(*lsf_r));
264 for (i = 0; i < LP_FILTER_ORDER; i++)
265 lsf_q[i] = lsf_r[i] * (LSF_R_FAC / 8000.0) + lsf_no_r[i] * (1.0 / 8000.0);
267 ff_set_min_dist_lsf(lsf_q, MIN_LSF_SPACING, LP_FILTER_ORDER);
270 interpolate_lsf(p->lsf_q, lsf_q);
272 ff_acelp_lsf2lspd(lsp, lsf_q, LP_FILTER_ORDER);
276 * Decode a set of 5 split-matrix quantized lsf indexes into 2 lsp vectors.
278 * @param p pointer to the AMRContext
280 static void lsf2lsp_5(AMRContext *p)
282 const uint16_t *lsf_param = p->frame.lsf;
283 float lsf_no_r[LP_FILTER_ORDER]; // LSFs without the residual vector
284 const int16_t *lsf_quantizer[5];
287 lsf_quantizer[0] = lsf_5_1[lsf_param[0]];
288 lsf_quantizer[1] = lsf_5_2[lsf_param[1]];
289 lsf_quantizer[2] = lsf_5_3[lsf_param[2] >> 1];
290 lsf_quantizer[3] = lsf_5_4[lsf_param[3]];
291 lsf_quantizer[4] = lsf_5_5[lsf_param[4]];
293 for (i = 0; i < LP_FILTER_ORDER; i++)
294 lsf_no_r[i] = p->prev_lsf_r[i] * LSF_R_FAC * PRED_FAC_MODE_12k2 + lsf_5_mean[i];
296 lsf2lsp_for_mode12k2(p, p->lsp[1], lsf_no_r, lsf_quantizer, 0, lsf_param[2] & 1, 0);
297 lsf2lsp_for_mode12k2(p, p->lsp[3], lsf_no_r, lsf_quantizer, 2, lsf_param[2] & 1, 1);
299 // interpolate LSP vectors at subframes 1 and 3
300 weighted_vector_sumd(p->lsp[0], p->prev_lsp_sub4, p->lsp[1], 0.5, 0.5, LP_FILTER_ORDER);
301 weighted_vector_sumd(p->lsp[2], p->lsp[1] , p->lsp[3], 0.5, 0.5, LP_FILTER_ORDER);
305 * Decode a set of 3 split-matrix quantized lsf indexes into an lsp vector.
307 * @param p pointer to the AMRContext
309 static void lsf2lsp_3(AMRContext *p)
311 const uint16_t *lsf_param = p->frame.lsf;
312 int16_t lsf_r[LP_FILTER_ORDER]; // residual LSF vector
313 float lsf_q[LP_FILTER_ORDER]; // quantified LSF vector
314 const int16_t *lsf_quantizer;
317 lsf_quantizer = (p->cur_frame_mode == MODE_7k95 ? lsf_3_1_MODE_7k95 : lsf_3_1)[lsf_param[0]];
318 memcpy(lsf_r, lsf_quantizer, 3 * sizeof(*lsf_r));
320 lsf_quantizer = lsf_3_2[lsf_param[1] << (p->cur_frame_mode <= MODE_5k15)];
321 memcpy(lsf_r + 3, lsf_quantizer, 3 * sizeof(*lsf_r));
323 lsf_quantizer = (p->cur_frame_mode <= MODE_5k15 ? lsf_3_3_MODE_5k15 : lsf_3_3)[lsf_param[2]];
324 memcpy(lsf_r + 6, lsf_quantizer, 4 * sizeof(*lsf_r));
326 // calculate mean-removed LSF vector and add mean
327 for (i = 0; i < LP_FILTER_ORDER; i++)
328 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);
330 ff_set_min_dist_lsf(lsf_q, MIN_LSF_SPACING, LP_FILTER_ORDER);
332 // store data for computing the next frame's LSFs
333 interpolate_lsf(p->lsf_q, lsf_q);
334 memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(*lsf_r));
336 ff_acelp_lsf2lspd(p->lsp[3], lsf_q, LP_FILTER_ORDER);
338 // interpolate LSP vectors at subframes 1, 2 and 3
339 for (i = 1; i <= 3; i++)
340 for(j = 0; j < LP_FILTER_ORDER; j++)
341 p->lsp[i-1][j] = p->prev_lsp_sub4[j] +
342 (p->lsp[3][j] - p->prev_lsp_sub4[j]) * 0.25 * i;
348 /// @name AMR pitch vector decoding functions
352 * Like ff_decode_pitch_lag(), but with 1/6 resolution
354 static void decode_pitch_lag_1_6(int *lag_int, int *lag_frac, int pitch_index,
355 const int prev_lag_int, const int subframe)
357 if (subframe == 0 || subframe == 2) {
358 if (pitch_index < 463) {
359 *lag_int = (pitch_index + 107) * 10923 >> 16;
360 *lag_frac = pitch_index - *lag_int * 6 + 105;
362 *lag_int = pitch_index - 368;
366 *lag_int = ((pitch_index + 5) * 10923 >> 16) - 1;
367 *lag_frac = pitch_index - *lag_int * 6 - 3;
368 *lag_int += av_clip(prev_lag_int - 5, PITCH_LAG_MIN_MODE_12k2,
369 PITCH_DELAY_MAX - 9);
373 static void decode_pitch_vector(AMRContext *p,
374 const AMRNBSubframe *amr_subframe,
377 int pitch_lag_int, pitch_lag_frac;
378 enum Mode mode = p->cur_frame_mode;
380 if (p->cur_frame_mode == MODE_12k2) {
381 decode_pitch_lag_1_6(&pitch_lag_int, &pitch_lag_frac,
382 amr_subframe->p_lag, p->pitch_lag_int,
385 ff_decode_pitch_lag(&pitch_lag_int, &pitch_lag_frac,
387 p->pitch_lag_int, subframe,
388 mode != MODE_4k75 && mode != MODE_5k15,
389 mode <= MODE_6k7 ? 4 : (mode == MODE_7k95 ? 5 : 6));
391 p->pitch_lag_int = pitch_lag_int; // store previous lag in a uint8_t
393 pitch_lag_frac <<= (p->cur_frame_mode != MODE_12k2);
395 pitch_lag_int += pitch_lag_frac > 0;
397 /* Calculate the pitch vector by interpolating the past excitation at the
398 pitch lag using a b60 hamming windowed sinc function. */
399 ff_acelp_interpolatef(p->excitation, p->excitation + 1 - pitch_lag_int,
401 pitch_lag_frac + 6 - 6*(pitch_lag_frac > 0),
402 10, AMR_SUBFRAME_SIZE);
404 memcpy(p->pitch_vector, p->excitation, AMR_SUBFRAME_SIZE * sizeof(float));
410 /// @name AMR algebraic code book (fixed) vector decoding functions
414 * Decode a 10-bit algebraic codebook index from a 10.2 kbit/s frame.
416 static void decode_10bit_pulse(int code, int pulse_position[8],
417 int i1, int i2, int i3)
419 // coded using 7+3 bits with the 3 LSBs being, individually, the LSB of 1 of
420 // the 3 pulses and the upper 7 bits being coded in base 5
421 const uint8_t *positions = base_five_table[code >> 3];
422 pulse_position[i1] = (positions[2] << 1) + ( code & 1);
423 pulse_position[i2] = (positions[1] << 1) + ((code >> 1) & 1);
424 pulse_position[i3] = (positions[0] << 1) + ((code >> 2) & 1);
428 * Decode the algebraic codebook index to pulse positions and signs and
429 * construct the algebraic codebook vector for MODE_10k2.
431 * @param fixed_index positions of the eight pulses
432 * @param fixed_sparse pointer to the algebraic codebook vector
434 static void decode_8_pulses_31bits(const int16_t *fixed_index,
435 AMRFixed *fixed_sparse)
437 int pulse_position[8];
440 decode_10bit_pulse(fixed_index[4], pulse_position, 0, 4, 1);
441 decode_10bit_pulse(fixed_index[5], pulse_position, 2, 6, 5);
443 // coded using 5+2 bits with the 2 LSBs being, individually, the LSB of 1 of
444 // the 2 pulses and the upper 5 bits being coded in base 5
445 temp = ((fixed_index[6] >> 2) * 25 + 12) >> 5;
446 pulse_position[3] = temp % 5;
447 pulse_position[7] = temp / 5;
448 if (pulse_position[7] & 1)
449 pulse_position[3] = 4 - pulse_position[3];
450 pulse_position[3] = (pulse_position[3] << 1) + ( fixed_index[6] & 1);
451 pulse_position[7] = (pulse_position[7] << 1) + ((fixed_index[6] >> 1) & 1);
454 for (i = 0; i < 4; i++) {
455 const int pos1 = (pulse_position[i] << 2) + i;
456 const int pos2 = (pulse_position[i + 4] << 2) + i;
457 const float sign = fixed_index[i] ? -1.0 : 1.0;
458 fixed_sparse->x[i ] = pos1;
459 fixed_sparse->x[i + 4] = pos2;
460 fixed_sparse->y[i ] = sign;
461 fixed_sparse->y[i + 4] = pos2 < pos1 ? -sign : sign;
466 * Decode the algebraic codebook index to pulse positions and signs,
467 * then construct the algebraic codebook vector.
469 * nb of pulses | bits encoding pulses
470 * For MODE_4k75 or MODE_5k15, 2 | 1-3, 4-6, 7
471 * MODE_5k9, 2 | 1, 2-4, 5-6, 7-9
472 * MODE_6k7, 3 | 1-3, 4, 5-7, 8, 9-11
473 * MODE_7k4 or MODE_7k95, 4 | 1-3, 4-6, 7-9, 10, 11-13
475 * @param fixed_sparse pointer to the algebraic codebook vector
476 * @param pulses algebraic codebook indexes
477 * @param mode mode of the current frame
478 * @param subframe current subframe number
480 static void decode_fixed_sparse(AMRFixed *fixed_sparse, const uint16_t *pulses,
481 const enum Mode mode, const int subframe)
483 assert(MODE_4k75 <= mode && mode <= MODE_12k2);
485 if (mode == MODE_12k2) {
486 ff_decode_10_pulses_35bits(pulses, fixed_sparse, gray_decode, 5, 3);
487 } else if (mode == MODE_10k2) {
488 decode_8_pulses_31bits(pulses, fixed_sparse);
490 int *pulse_position = fixed_sparse->x;
492 const int fixed_index = pulses[0];
494 if (mode <= MODE_5k15) {
495 pulse_subset = ((fixed_index >> 3) & 8) + (subframe << 1);
496 pulse_position[0] = ( fixed_index & 7) * 5 + track_position[pulse_subset];
497 pulse_position[1] = ((fixed_index >> 3) & 7) * 5 + track_position[pulse_subset + 1];
499 } else if (mode == MODE_5k9) {
500 pulse_subset = ((fixed_index & 1) << 1) + 1;
501 pulse_position[0] = ((fixed_index >> 1) & 7) * 5 + pulse_subset;
502 pulse_subset = (fixed_index >> 4) & 3;
503 pulse_position[1] = ((fixed_index >> 6) & 7) * 5 + pulse_subset + (pulse_subset == 3 ? 1 : 0);
504 fixed_sparse->n = pulse_position[0] == pulse_position[1] ? 1 : 2;
505 } else if (mode == MODE_6k7) {
506 pulse_position[0] = (fixed_index & 7) * 5;
507 pulse_subset = (fixed_index >> 2) & 2;
508 pulse_position[1] = ((fixed_index >> 4) & 7) * 5 + pulse_subset + 1;
509 pulse_subset = (fixed_index >> 6) & 2;
510 pulse_position[2] = ((fixed_index >> 8) & 7) * 5 + pulse_subset + 2;
512 } else { // mode <= MODE_7k95
513 pulse_position[0] = gray_decode[ fixed_index & 7];
514 pulse_position[1] = gray_decode[(fixed_index >> 3) & 7] + 1;
515 pulse_position[2] = gray_decode[(fixed_index >> 6) & 7] + 2;
516 pulse_subset = (fixed_index >> 9) & 1;
517 pulse_position[3] = gray_decode[(fixed_index >> 10) & 7] + pulse_subset + 3;
520 for (i = 0; i < fixed_sparse->n; i++)
521 fixed_sparse->y[i] = (pulses[1] >> i) & 1 ? 1.0 : -1.0;
526 * Apply pitch lag to obtain the sharpened fixed vector (section 6.1.2)
528 * @param p the context
529 * @param subframe unpacked amr subframe
530 * @param mode mode of the current frame
531 * @param fixed_sparse sparse respresentation of the fixed vector
533 static void pitch_sharpening(AMRContext *p, int subframe, enum Mode mode,
534 AMRFixed *fixed_sparse)
536 // The spec suggests the current pitch gain is always used, but in other
537 // modes the pitch and codebook gains are joinly quantized (sec 5.8.2)
538 // so the codebook gain cannot depend on the quantized pitch gain.
539 if (mode == MODE_12k2)
540 p->beta = FFMIN(p->pitch_gain[4], 1.0);
542 fixed_sparse->pitch_lag = p->pitch_lag_int;
543 fixed_sparse->pitch_fac = p->beta;
545 // Save pitch sharpening factor for the next subframe
546 // MODE_4k75 only updates on the 2nd and 4th subframes - this follows from
547 // the fact that the gains for two subframes are jointly quantized.
548 if (mode != MODE_4k75 || subframe & 1)
549 p->beta = av_clipf(p->pitch_gain[4], 0.0, SHARP_MAX);
554 /// @name AMR gain decoding functions
558 * fixed gain smoothing
559 * Note that where the spec specifies the "spectrum in the q domain"
560 * in section 6.1.4, in fact frequencies should be used.
562 * @param p the context
563 * @param lsf LSFs for the current subframe, in the range [0,1]
564 * @param lsf_avg averaged LSFs
565 * @param mode mode of the current frame
567 * @return fixed gain smoothed
569 static float fixed_gain_smooth(AMRContext *p , const float *lsf,
570 const float *lsf_avg, const enum Mode mode)
575 for (i = 0; i < LP_FILTER_ORDER; i++)
576 diff += fabs(lsf_avg[i] - lsf[i]) / lsf_avg[i];
578 // If diff is large for ten subframes, disable smoothing for a 40-subframe
584 if (p->diff_count > 10) {
586 p->diff_count--; // don't let diff_count overflow
589 if (p->hang_count < 40) {
591 } else if (mode < MODE_7k4 || mode == MODE_10k2) {
592 const float smoothing_factor = av_clipf(4.0 * diff - 1.6, 0.0, 1.0);
593 const float fixed_gain_mean = (p->fixed_gain[0] + p->fixed_gain[1] +
594 p->fixed_gain[2] + p->fixed_gain[3] +
595 p->fixed_gain[4]) * 0.2;
596 return smoothing_factor * p->fixed_gain[4] +
597 (1.0 - smoothing_factor) * fixed_gain_mean;
599 return p->fixed_gain[4];
603 * Decode pitch gain and fixed gain factor (part of section 6.1.3).
605 * @param p the context
606 * @param amr_subframe unpacked amr subframe
607 * @param mode mode of the current frame
608 * @param subframe current subframe number
609 * @param fixed_gain_factor decoded gain correction factor
611 static void decode_gains(AMRContext *p, const AMRNBSubframe *amr_subframe,
612 const enum Mode mode, const int subframe,
613 float *fixed_gain_factor)
615 if (mode == MODE_12k2 || mode == MODE_7k95) {
616 p->pitch_gain[4] = qua_gain_pit [amr_subframe->p_gain ]
618 *fixed_gain_factor = qua_gain_code[amr_subframe->fixed_gain]
621 const uint16_t *gains;
623 if (mode >= MODE_6k7) {
624 gains = gains_high[amr_subframe->p_gain];
625 } else if (mode >= MODE_5k15) {
626 gains = gains_low [amr_subframe->p_gain];
628 // gain index is only coded in subframes 0,2 for MODE_4k75
629 gains = gains_MODE_4k75[(p->frame.subframe[subframe & 2].p_gain << 1) + (subframe & 1)];
632 p->pitch_gain[4] = gains[0] * (1.0 / 16384.0);
633 *fixed_gain_factor = gains[1] * (1.0 / 4096.0);
640 /// @name AMR preprocessing functions
644 * Circularly convolve a sparse fixed vector with a phase dispersion impulse
645 * response filter (D.6.2 of G.729 and 6.1.5 of AMR).
647 * @param out vector with filter applied
648 * @param in source vector
649 * @param filter phase filter coefficients
651 * out[n] = sum(i,0,len-1){ in[i] * filter[(len + n - i)%len] }
653 static void apply_ir_filter(float *out, const AMRFixed *in,
656 float filter1[AMR_SUBFRAME_SIZE], ///< filters at pitch lag*1 and *2
657 filter2[AMR_SUBFRAME_SIZE];
658 int lag = in->pitch_lag;
659 float fac = in->pitch_fac;
662 if (lag < AMR_SUBFRAME_SIZE) {
663 ff_celp_circ_addf(filter1, filter, filter, lag, fac,
666 if (lag < AMR_SUBFRAME_SIZE >> 1)
667 ff_celp_circ_addf(filter2, filter, filter1, lag, fac,
671 memset(out, 0, sizeof(float) * AMR_SUBFRAME_SIZE);
672 for (i = 0; i < in->n; i++) {
675 const float *filterp;
677 if (x >= AMR_SUBFRAME_SIZE - lag) {
679 } else if (x >= AMR_SUBFRAME_SIZE - (lag << 1)) {
684 ff_celp_circ_addf(out, out, filterp, x, y, AMR_SUBFRAME_SIZE);
689 * Reduce fixed vector sparseness by smoothing with one of three IR filters.
690 * Also know as "adaptive phase dispersion".
692 * This implements 3GPP TS 26.090 section 6.1(5).
694 * @param p the context
695 * @param fixed_sparse algebraic codebook vector
696 * @param fixed_vector unfiltered fixed vector
697 * @param fixed_gain smoothed gain
698 * @param out space for modified vector if necessary
700 static const float *anti_sparseness(AMRContext *p, AMRFixed *fixed_sparse,
701 const float *fixed_vector,
702 float fixed_gain, float *out)
706 if (p->pitch_gain[4] < 0.6) {
707 ir_filter_nr = 0; // strong filtering
708 } else if (p->pitch_gain[4] < 0.9) {
709 ir_filter_nr = 1; // medium filtering
711 ir_filter_nr = 2; // no filtering
714 if (fixed_gain > 2.0 * p->prev_sparse_fixed_gain) {
715 p->ir_filter_onset = 2;
716 } else if (p->ir_filter_onset)
717 p->ir_filter_onset--;
719 if (!p->ir_filter_onset) {
722 for (i = 0; i < 5; i++)
723 if (p->pitch_gain[i] < 0.6)
728 if (ir_filter_nr > p->prev_ir_filter_nr + 1)
730 } else if (ir_filter_nr < 2)
733 // Disable filtering for very low level of fixed_gain.
734 // Note this step is not specified in the technical description but is in
735 // the reference source in the function Ph_disp.
736 if (fixed_gain < 5.0)
739 if (p->cur_frame_mode != MODE_7k4 && p->cur_frame_mode < MODE_10k2
740 && ir_filter_nr < 2) {
741 apply_ir_filter(out, fixed_sparse,
742 (p->cur_frame_mode == MODE_7k95 ?
743 ir_filters_lookup_MODE_7k95 :
744 ir_filters_lookup)[ir_filter_nr]);
748 // update ir filter strength history
749 p->prev_ir_filter_nr = ir_filter_nr;
750 p->prev_sparse_fixed_gain = fixed_gain;
758 /// @name AMR synthesis functions
762 * Conduct 10th order linear predictive coding synthesis.
764 * @param p pointer to the AMRContext
765 * @param lpc pointer to the LPC coefficients
766 * @param fixed_gain fixed codebook gain for synthesis
767 * @param fixed_vector algebraic codebook vector
768 * @param samples pointer to the output speech samples
769 * @param overflow 16-bit overflow flag
771 static int synthesis(AMRContext *p, float *lpc,
772 float fixed_gain, const float *fixed_vector,
773 float *samples, uint8_t overflow)
776 float excitation[AMR_SUBFRAME_SIZE];
778 // if an overflow has been detected, the pitch vector is scaled down by a
781 for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
782 p->pitch_vector[i] *= 0.25;
784 ff_weighted_vector_sumf(excitation, p->pitch_vector, fixed_vector,
785 p->pitch_gain[4], fixed_gain, AMR_SUBFRAME_SIZE);
787 // emphasize pitch vector contribution
788 if (p->pitch_gain[4] > 0.5 && !overflow) {
789 float energy = ff_dot_productf(excitation, excitation,
793 (p->cur_frame_mode == MODE_12k2 ?
794 0.25 * FFMIN(p->pitch_gain[4], 1.0) :
795 0.5 * FFMIN(p->pitch_gain[4], SHARP_MAX));
797 for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
798 excitation[i] += pitch_factor * p->pitch_vector[i];
800 ff_scale_vector_to_given_sum_of_squares(excitation, excitation, energy,
804 ff_celp_lp_synthesis_filterf(samples, lpc, excitation, AMR_SUBFRAME_SIZE,
808 for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
809 if (fabsf(samples[i]) > AMR_SAMPLE_BOUND) {
819 /// @name AMR update functions
823 * Update buffers and history at the end of decoding a subframe.
825 * @param p pointer to the AMRContext
827 static void update_state(AMRContext *p)
829 memcpy(p->prev_lsp_sub4, p->lsp[3], LP_FILTER_ORDER * sizeof(p->lsp[3][0]));
831 memmove(&p->excitation_buf[0], &p->excitation_buf[AMR_SUBFRAME_SIZE],
832 (PITCH_DELAY_MAX + LP_FILTER_ORDER + 1) * sizeof(float));
834 memmove(&p->pitch_gain[0], &p->pitch_gain[1], 4 * sizeof(float));
835 memmove(&p->fixed_gain[0], &p->fixed_gain[1], 4 * sizeof(float));
837 memmove(&p->samples_in[0], &p->samples_in[AMR_SUBFRAME_SIZE],
838 LP_FILTER_ORDER * sizeof(float));
844 /// @name AMR Postprocessing functions
848 * Get the tilt factor of a formant filter from its transfer function
850 * @param lpc_n LP_FILTER_ORDER coefficients of the numerator
851 * @param lpc_d LP_FILTER_ORDER coefficients of the denominator
853 static float tilt_factor(float *lpc_n, float *lpc_d)
855 float rh0, rh1; // autocorrelation at lag 0 and 1
857 // LP_FILTER_ORDER prior zeros are needed for ff_celp_lp_synthesis_filterf
858 float impulse_buffer[LP_FILTER_ORDER + AMR_TILT_RESPONSE] = { 0 };
859 float *hf = impulse_buffer + LP_FILTER_ORDER; // start of impulse response
862 memcpy(hf + 1, lpc_n, sizeof(float) * LP_FILTER_ORDER);
863 ff_celp_lp_synthesis_filterf(hf, lpc_d, hf, AMR_TILT_RESPONSE,
866 rh0 = ff_dot_productf(hf, hf, AMR_TILT_RESPONSE);
867 rh1 = ff_dot_productf(hf, hf + 1, AMR_TILT_RESPONSE - 1);
869 // The spec only specifies this check for 12.2 and 10.2 kbit/s
870 // modes. But in the ref source the tilt is always non-negative.
871 return rh1 >= 0.0 ? rh1 / rh0 * AMR_TILT_GAMMA_T : 0.0;
875 * Perform adaptive post-filtering to enhance the quality of the speech.
878 * @param p pointer to the AMRContext
879 * @param lpc interpolated LP coefficients for this subframe
880 * @param buf_out output of the filter
882 static void postfilter(AMRContext *p, float *lpc, float *buf_out)
885 float *samples = p->samples_in + LP_FILTER_ORDER; // Start of input
887 float speech_gain = ff_dot_productf(samples, samples,
890 float pole_out[AMR_SUBFRAME_SIZE + LP_FILTER_ORDER]; // Output of pole filter
891 const float *gamma_n, *gamma_d; // Formant filter factor table
892 float lpc_n[LP_FILTER_ORDER], lpc_d[LP_FILTER_ORDER]; // Transfer function coefficients
894 if (p->cur_frame_mode == MODE_12k2 || p->cur_frame_mode == MODE_10k2) {
895 gamma_n = ff_pow_0_7;
896 gamma_d = ff_pow_0_75;
898 gamma_n = ff_pow_0_55;
899 gamma_d = ff_pow_0_7;
902 for (i = 0; i < LP_FILTER_ORDER; i++) {
903 lpc_n[i] = lpc[i] * gamma_n[i];
904 lpc_d[i] = lpc[i] * gamma_d[i];
907 memcpy(pole_out, p->postfilter_mem, sizeof(float) * LP_FILTER_ORDER);
908 ff_celp_lp_synthesis_filterf(pole_out + LP_FILTER_ORDER, lpc_d, samples,
909 AMR_SUBFRAME_SIZE, LP_FILTER_ORDER);
910 memcpy(p->postfilter_mem, pole_out + AMR_SUBFRAME_SIZE,
911 sizeof(float) * LP_FILTER_ORDER);
913 ff_celp_lp_zero_synthesis_filterf(buf_out, lpc_n,
914 pole_out + LP_FILTER_ORDER,
915 AMR_SUBFRAME_SIZE, LP_FILTER_ORDER);
917 ff_tilt_compensation(&p->tilt_mem, tilt_factor(lpc_n, lpc_d), buf_out,
920 ff_adaptive_gain_control(buf_out, buf_out, speech_gain, AMR_SUBFRAME_SIZE,
921 AMR_AGC_ALPHA, &p->postfilter_agc);
926 static int amrnb_decode_frame(AVCodecContext *avctx, void *data,
927 int *got_frame_ptr, AVPacket *avpkt)
930 AMRContext *p = avctx->priv_data; // pointer to private data
931 const uint8_t *buf = avpkt->data;
932 int buf_size = avpkt->size;
933 float *buf_out; // pointer to the output data buffer
934 int i, subframe, ret;
935 float fixed_gain_factor;
936 AMRFixed fixed_sparse = {0}; // fixed vector up to anti-sparseness processing
937 float spare_vector[AMR_SUBFRAME_SIZE]; // extra stack space to hold result from anti-sparseness processing
938 float synth_fixed_gain; // the fixed gain that synthesis should use
939 const float *synth_fixed_vector; // pointer to the fixed vector that synthesis should use
941 /* get output buffer */
942 p->avframe.nb_samples = AMR_BLOCK_SIZE;
943 if ((ret = avctx->get_buffer(avctx, &p->avframe)) < 0) {
944 av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
947 buf_out = (float *)p->avframe.data[0];
949 p->cur_frame_mode = unpack_bitstream(p, buf, buf_size);
950 if (p->cur_frame_mode == MODE_DTX) {
951 av_log_missing_feature(avctx, "dtx mode", 0);
952 av_log(avctx, AV_LOG_INFO, "Note: libopencore_amrnb supports dtx\n");
956 if (p->cur_frame_mode == MODE_12k2) {
961 for (i = 0; i < 4; i++)
962 ff_acelp_lspd2lpc(p->lsp[i], p->lpc[i], 5);
964 for (subframe = 0; subframe < 4; subframe++) {
965 const AMRNBSubframe *amr_subframe = &p->frame.subframe[subframe];
967 decode_pitch_vector(p, amr_subframe, subframe);
969 decode_fixed_sparse(&fixed_sparse, amr_subframe->pulses,
970 p->cur_frame_mode, subframe);
972 // The fixed gain (section 6.1.3) depends on the fixed vector
973 // (section 6.1.2), but the fixed vector calculation uses
974 // pitch sharpening based on the on the pitch gain (section 6.1.3).
975 // So the correct order is: pitch gain, pitch sharpening, fixed gain.
976 decode_gains(p, amr_subframe, p->cur_frame_mode, subframe,
979 pitch_sharpening(p, subframe, p->cur_frame_mode, &fixed_sparse);
981 if (fixed_sparse.pitch_lag == 0) {
982 av_log(avctx, AV_LOG_ERROR, "The file is corrupted, pitch_lag = 0 is not allowed\n");
983 return AVERROR_INVALIDDATA;
985 ff_set_fixed_vector(p->fixed_vector, &fixed_sparse, 1.0,
989 ff_amr_set_fixed_gain(fixed_gain_factor,
990 ff_dot_productf(p->fixed_vector, p->fixed_vector,
991 AMR_SUBFRAME_SIZE)/AMR_SUBFRAME_SIZE,
993 energy_mean[p->cur_frame_mode], energy_pred_fac);
995 // The excitation feedback is calculated without any processing such
996 // as fixed gain smoothing. This isn't mentioned in the specification.
997 for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
998 p->excitation[i] *= p->pitch_gain[4];
999 ff_set_fixed_vector(p->excitation, &fixed_sparse, p->fixed_gain[4],
1002 // In the ref decoder, excitation is stored with no fractional bits.
1003 // This step prevents buzz in silent periods. The ref encoder can
1004 // emit long sequences with pitch factor greater than one. This
1005 // creates unwanted feedback if the excitation vector is nonzero.
1006 // (e.g. test sequence T19_795.COD in 3GPP TS 26.074)
1007 for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
1008 p->excitation[i] = truncf(p->excitation[i]);
1010 // Smooth fixed gain.
1011 // The specification is ambiguous, but in the reference source, the
1012 // smoothed value is NOT fed back into later fixed gain smoothing.
1013 synth_fixed_gain = fixed_gain_smooth(p, p->lsf_q[subframe],
1014 p->lsf_avg, p->cur_frame_mode);
1016 synth_fixed_vector = anti_sparseness(p, &fixed_sparse, p->fixed_vector,
1017 synth_fixed_gain, spare_vector);
1019 if (synthesis(p, p->lpc[subframe], synth_fixed_gain,
1020 synth_fixed_vector, &p->samples_in[LP_FILTER_ORDER], 0))
1021 // overflow detected -> rerun synthesis scaling pitch vector down
1022 // by a factor of 4, skipping pitch vector contribution emphasis
1023 // and adaptive gain control
1024 synthesis(p, p->lpc[subframe], synth_fixed_gain,
1025 synth_fixed_vector, &p->samples_in[LP_FILTER_ORDER], 1);
1027 postfilter(p, p->lpc[subframe], buf_out + subframe * AMR_SUBFRAME_SIZE);
1029 // update buffers and history
1030 ff_clear_fixed_vector(p->fixed_vector, &fixed_sparse, AMR_SUBFRAME_SIZE);
1034 ff_acelp_apply_order_2_transfer_function(buf_out, buf_out, highpass_zeros,
1036 highpass_gain * AMR_SAMPLE_SCALE,
1037 p->high_pass_mem, AMR_BLOCK_SIZE);
1039 /* Update averaged lsf vector (used for fixed gain smoothing).
1041 * Note that lsf_avg should not incorporate the current frame's LSFs
1042 * for fixed_gain_smooth.
1043 * The specification has an incorrect formula: the reference decoder uses
1044 * qbar(n-1) rather than qbar(n) in section 6.1(4) equation 71. */
1045 ff_weighted_vector_sumf(p->lsf_avg, p->lsf_avg, p->lsf_q[3],
1046 0.84, 0.16, LP_FILTER_ORDER);
1049 *(AVFrame *)data = p->avframe;
1051 /* return the amount of bytes consumed if everything was OK */
1052 return frame_sizes_nb[p->cur_frame_mode] + 1; // +7 for rounding and +8 for TOC
1056 AVCodec ff_amrnb_decoder = {
1058 .type = AVMEDIA_TYPE_AUDIO,
1059 .id = CODEC_ID_AMR_NB,
1060 .priv_data_size = sizeof(AMRContext),
1061 .init = amrnb_decode_init,
1062 .decode = amrnb_decode_frame,
1063 .capabilities = CODEC_CAP_DR1,
1064 .long_name = NULL_IF_CONFIG_SMALL("Adaptive Multi-Rate NarrowBand"),
1065 .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE},