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
46 #include "libavutil/channel_layout.h"
49 #include "libavutil/common.h"
50 #include "libavutil/avassert.h"
51 #include "celp_math.h"
52 #include "celp_filters.h"
53 #include "acelp_filters.h"
54 #include "acelp_vectors.h"
55 #include "acelp_pitch_delay.h"
60 #include "amrnbdata.h"
62 #define AMR_BLOCK_SIZE 160 ///< samples per frame
63 #define AMR_SAMPLE_BOUND 32768.0 ///< threshold for synthesis overflow
66 * Scale from constructed speech to [-1,1]
68 * AMR is designed to produce 16-bit PCM samples (3GPP TS 26.090 4.2) but
69 * upscales by two (section 6.2.2).
71 * Fundamentally, this scale is determined by energy_mean through
72 * the fixed vector contribution to the excitation vector.
74 #define AMR_SAMPLE_SCALE (2.0 / 32768.0)
76 /** Prediction factor for 12.2kbit/s mode */
77 #define PRED_FAC_MODE_12k2 0.65
79 #define LSF_R_FAC (8000.0 / 32768.0) ///< LSF residual tables to Hertz
80 #define MIN_LSF_SPACING (50.0488 / 8000.0) ///< Ensures stability of LPC filter
81 #define PITCH_LAG_MIN_MODE_12k2 18 ///< Lower bound on decoded lag search in 12.2kbit/s mode
83 /** Initial energy in dB. Also used for bad frames (unimplemented). */
84 #define MIN_ENERGY -14.0
86 /** Maximum sharpening factor
88 * The specification says 0.8, which should be 13107, but the reference C code
89 * uses 13017 instead. (Amusingly the same applies to SHARP_MAX in g729dec.c.)
91 #define SHARP_MAX 0.79449462890625
93 /** Number of impulse response coefficients used for tilt factor */
94 #define AMR_TILT_RESPONSE 22
95 /** Tilt factor = 1st reflection coefficient * gamma_t */
96 #define AMR_TILT_GAMMA_T 0.8
97 /** Adaptive gain control factor used in post-filter */
98 #define AMR_AGC_ALPHA 0.9
100 typedef struct AMRContext {
101 AVFrame avframe; ///< AVFrame for decoded samples
102 AMRNBFrame frame; ///< decoded AMR parameters (lsf coefficients, codebook indexes, etc)
103 uint8_t bad_frame_indicator; ///< bad frame ? 1 : 0
104 enum Mode cur_frame_mode;
106 int16_t prev_lsf_r[LP_FILTER_ORDER]; ///< residual LSF vector from previous subframe
107 double lsp[4][LP_FILTER_ORDER]; ///< lsp vectors from current frame
108 double prev_lsp_sub4[LP_FILTER_ORDER]; ///< lsp vector for the 4th subframe of the previous frame
110 float lsf_q[4][LP_FILTER_ORDER]; ///< Interpolated LSF vector for fixed gain smoothing
111 float lsf_avg[LP_FILTER_ORDER]; ///< vector of averaged lsf vector
113 float lpc[4][LP_FILTER_ORDER]; ///< lpc coefficient vectors for 4 subframes
115 uint8_t pitch_lag_int; ///< integer part of pitch lag from current subframe
117 float excitation_buf[PITCH_DELAY_MAX + LP_FILTER_ORDER + 1 + AMR_SUBFRAME_SIZE]; ///< current excitation and all necessary excitation history
118 float *excitation; ///< pointer to the current excitation vector in excitation_buf
120 float pitch_vector[AMR_SUBFRAME_SIZE]; ///< adaptive code book (pitch) vector
121 float fixed_vector[AMR_SUBFRAME_SIZE]; ///< algebraic codebook (fixed) vector (must be kept zero between frames)
123 float prediction_error[4]; ///< quantified prediction errors {20log10(^gamma_gc)} for previous four subframes
124 float pitch_gain[5]; ///< quantified pitch gains for the current and previous four subframes
125 float fixed_gain[5]; ///< quantified fixed gains for the current and previous four subframes
127 float beta; ///< previous pitch_gain, bounded by [0.0,SHARP_MAX]
128 uint8_t diff_count; ///< the number of subframes for which diff has been above 0.65
129 uint8_t hang_count; ///< the number of subframes since a hangover period started
131 float prev_sparse_fixed_gain; ///< previous fixed gain; used by anti-sparseness processing to determine "onset"
132 uint8_t prev_ir_filter_nr; ///< previous impulse response filter "impNr": 0 - strong, 1 - medium, 2 - none
133 uint8_t ir_filter_onset; ///< flag for impulse response filter strength
135 float postfilter_mem[10]; ///< previous intermediate values in the formant filter
136 float tilt_mem; ///< previous input to tilt compensation filter
137 float postfilter_agc; ///< previous factor used for adaptive gain control
138 float high_pass_mem[2]; ///< previous intermediate values in the high-pass filter
140 float samples_in[LP_FILTER_ORDER + AMR_SUBFRAME_SIZE]; ///< floating point samples
142 ACELPFContext acelpf_ctx; ///< context for filters for ACELP-based codecs
143 ACELPVContext acelpv_ctx; ///< context for vector operations for ACELP-based codecs
144 CELPFContext celpf_ctx; ///< context for filters for CELP-based codecs
145 CELPMContext celpm_ctx; ///< context for fixed point math operations
149 /** Double version of ff_weighted_vector_sumf() */
150 static void weighted_vector_sumd(double *out, const double *in_a,
151 const double *in_b, double weight_coeff_a,
152 double weight_coeff_b, int length)
156 for (i = 0; i < length; i++)
157 out[i] = weight_coeff_a * in_a[i]
158 + weight_coeff_b * in_b[i];
161 static av_cold int amrnb_decode_init(AVCodecContext *avctx)
163 AMRContext *p = avctx->priv_data;
166 if (avctx->channels > 1) {
167 av_log_missing_feature(avctx, "multi-channel AMR", 0);
168 return AVERROR_PATCHWELCOME;
172 avctx->channel_layout = AV_CH_LAYOUT_MONO;
173 if (!avctx->sample_rate)
174 avctx->sample_rate = 8000;
175 avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
177 // p->excitation always points to the same position in p->excitation_buf
178 p->excitation = &p->excitation_buf[PITCH_DELAY_MAX + LP_FILTER_ORDER + 1];
180 for (i = 0; i < LP_FILTER_ORDER; i++) {
181 p->prev_lsp_sub4[i] = lsp_sub4_init[i] * 1000 / (float)(1 << 15);
182 p->lsf_avg[i] = p->lsf_q[3][i] = lsp_avg_init[i] / (float)(1 << 15);
185 for (i = 0; i < 4; i++)
186 p->prediction_error[i] = MIN_ENERGY;
188 avcodec_get_frame_defaults(&p->avframe);
189 avctx->coded_frame = &p->avframe;
191 ff_acelp_filter_init(&p->acelpf_ctx);
192 ff_acelp_vectors_init(&p->acelpv_ctx);
193 ff_celp_filter_init(&p->celpf_ctx);
194 ff_celp_math_init(&p->celpm_ctx);
201 * Unpack an RFC4867 speech frame into the AMR frame mode and parameters.
203 * The order of speech bits is specified by 3GPP TS 26.101.
205 * @param p the context
206 * @param buf pointer to the input buffer
207 * @param buf_size size of the input buffer
209 * @return the frame mode
211 static enum Mode unpack_bitstream(AMRContext *p, const uint8_t *buf,
216 // Decode the first octet.
217 mode = buf[0] >> 3 & 0x0F; // frame type
218 p->bad_frame_indicator = (buf[0] & 0x4) != 0x4; // quality bit
220 if (mode >= N_MODES || buf_size < frame_sizes_nb[mode] + 1) {
225 ff_amr_bit_reorder((uint16_t *) &p->frame, sizeof(AMRNBFrame), buf + 1,
226 amr_unpacking_bitmaps_per_mode[mode]);
232 /// @name AMR pitch LPC coefficient decoding functions
236 * Interpolate the LSF vector (used for fixed gain smoothing).
237 * The interpolation is done over all four subframes even in MODE_12k2.
239 * @param[in] ctx The Context
240 * @param[in,out] lsf_q LSFs in [0,1] for each subframe
241 * @param[in] lsf_new New LSFs in [0,1] for subframe 4
243 static void interpolate_lsf(ACELPVContext *ctx, float lsf_q[4][LP_FILTER_ORDER], float *lsf_new)
247 for (i = 0; i < 4; i++)
248 ctx->weighted_vector_sumf(lsf_q[i], lsf_q[3], lsf_new,
249 0.25 * (3 - i), 0.25 * (i + 1),
254 * Decode a set of 5 split-matrix quantized lsf indexes into an lsp vector.
256 * @param p the context
257 * @param lsp output LSP vector
258 * @param lsf_no_r LSF vector without the residual vector added
259 * @param lsf_quantizer pointers to LSF dictionary tables
260 * @param quantizer_offset offset in tables
261 * @param sign for the 3 dictionary table
262 * @param update store data for computing the next frame's LSFs
264 static void lsf2lsp_for_mode12k2(AMRContext *p, double lsp[LP_FILTER_ORDER],
265 const float lsf_no_r[LP_FILTER_ORDER],
266 const int16_t *lsf_quantizer[5],
267 const int quantizer_offset,
268 const int sign, const int update)
270 int16_t lsf_r[LP_FILTER_ORDER]; // residual LSF vector
271 float lsf_q[LP_FILTER_ORDER]; // quantified LSF vector
274 for (i = 0; i < LP_FILTER_ORDER >> 1; i++)
275 memcpy(&lsf_r[i << 1], &lsf_quantizer[i][quantizer_offset],
284 memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(*lsf_r));
286 for (i = 0; i < LP_FILTER_ORDER; i++)
287 lsf_q[i] = lsf_r[i] * (LSF_R_FAC / 8000.0) + lsf_no_r[i] * (1.0 / 8000.0);
289 ff_set_min_dist_lsf(lsf_q, MIN_LSF_SPACING, LP_FILTER_ORDER);
292 interpolate_lsf(&p->acelpv_ctx, p->lsf_q, lsf_q);
294 ff_acelp_lsf2lspd(lsp, lsf_q, LP_FILTER_ORDER);
298 * Decode a set of 5 split-matrix quantized lsf indexes into 2 lsp vectors.
300 * @param p pointer to the AMRContext
302 static void lsf2lsp_5(AMRContext *p)
304 const uint16_t *lsf_param = p->frame.lsf;
305 float lsf_no_r[LP_FILTER_ORDER]; // LSFs without the residual vector
306 const int16_t *lsf_quantizer[5];
309 lsf_quantizer[0] = lsf_5_1[lsf_param[0]];
310 lsf_quantizer[1] = lsf_5_2[lsf_param[1]];
311 lsf_quantizer[2] = lsf_5_3[lsf_param[2] >> 1];
312 lsf_quantizer[3] = lsf_5_4[lsf_param[3]];
313 lsf_quantizer[4] = lsf_5_5[lsf_param[4]];
315 for (i = 0; i < LP_FILTER_ORDER; i++)
316 lsf_no_r[i] = p->prev_lsf_r[i] * LSF_R_FAC * PRED_FAC_MODE_12k2 + lsf_5_mean[i];
318 lsf2lsp_for_mode12k2(p, p->lsp[1], lsf_no_r, lsf_quantizer, 0, lsf_param[2] & 1, 0);
319 lsf2lsp_for_mode12k2(p, p->lsp[3], lsf_no_r, lsf_quantizer, 2, lsf_param[2] & 1, 1);
321 // interpolate LSP vectors at subframes 1 and 3
322 weighted_vector_sumd(p->lsp[0], p->prev_lsp_sub4, p->lsp[1], 0.5, 0.5, LP_FILTER_ORDER);
323 weighted_vector_sumd(p->lsp[2], p->lsp[1] , p->lsp[3], 0.5, 0.5, LP_FILTER_ORDER);
327 * Decode a set of 3 split-matrix quantized lsf indexes into an lsp vector.
329 * @param p pointer to the AMRContext
331 static void lsf2lsp_3(AMRContext *p)
333 const uint16_t *lsf_param = p->frame.lsf;
334 int16_t lsf_r[LP_FILTER_ORDER]; // residual LSF vector
335 float lsf_q[LP_FILTER_ORDER]; // quantified LSF vector
336 const int16_t *lsf_quantizer;
339 lsf_quantizer = (p->cur_frame_mode == MODE_7k95 ? lsf_3_1_MODE_7k95 : lsf_3_1)[lsf_param[0]];
340 memcpy(lsf_r, lsf_quantizer, 3 * sizeof(*lsf_r));
342 lsf_quantizer = lsf_3_2[lsf_param[1] << (p->cur_frame_mode <= MODE_5k15)];
343 memcpy(lsf_r + 3, lsf_quantizer, 3 * sizeof(*lsf_r));
345 lsf_quantizer = (p->cur_frame_mode <= MODE_5k15 ? lsf_3_3_MODE_5k15 : lsf_3_3)[lsf_param[2]];
346 memcpy(lsf_r + 6, lsf_quantizer, 4 * sizeof(*lsf_r));
348 // calculate mean-removed LSF vector and add mean
349 for (i = 0; i < LP_FILTER_ORDER; i++)
350 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);
352 ff_set_min_dist_lsf(lsf_q, MIN_LSF_SPACING, LP_FILTER_ORDER);
354 // store data for computing the next frame's LSFs
355 interpolate_lsf(&p->acelpv_ctx, p->lsf_q, lsf_q);
356 memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(*lsf_r));
358 ff_acelp_lsf2lspd(p->lsp[3], lsf_q, LP_FILTER_ORDER);
360 // interpolate LSP vectors at subframes 1, 2 and 3
361 for (i = 1; i <= 3; i++)
362 for(j = 0; j < LP_FILTER_ORDER; j++)
363 p->lsp[i-1][j] = p->prev_lsp_sub4[j] +
364 (p->lsp[3][j] - p->prev_lsp_sub4[j]) * 0.25 * i;
370 /// @name AMR pitch vector decoding functions
374 * Like ff_decode_pitch_lag(), but with 1/6 resolution
376 static void decode_pitch_lag_1_6(int *lag_int, int *lag_frac, int pitch_index,
377 const int prev_lag_int, const int subframe)
379 if (subframe == 0 || subframe == 2) {
380 if (pitch_index < 463) {
381 *lag_int = (pitch_index + 107) * 10923 >> 16;
382 *lag_frac = pitch_index - *lag_int * 6 + 105;
384 *lag_int = pitch_index - 368;
388 *lag_int = ((pitch_index + 5) * 10923 >> 16) - 1;
389 *lag_frac = pitch_index - *lag_int * 6 - 3;
390 *lag_int += av_clip(prev_lag_int - 5, PITCH_LAG_MIN_MODE_12k2,
391 PITCH_DELAY_MAX - 9);
395 static void decode_pitch_vector(AMRContext *p,
396 const AMRNBSubframe *amr_subframe,
399 int pitch_lag_int, pitch_lag_frac;
400 enum Mode mode = p->cur_frame_mode;
402 if (p->cur_frame_mode == MODE_12k2) {
403 decode_pitch_lag_1_6(&pitch_lag_int, &pitch_lag_frac,
404 amr_subframe->p_lag, p->pitch_lag_int,
407 ff_decode_pitch_lag(&pitch_lag_int, &pitch_lag_frac,
409 p->pitch_lag_int, subframe,
410 mode != MODE_4k75 && mode != MODE_5k15,
411 mode <= MODE_6k7 ? 4 : (mode == MODE_7k95 ? 5 : 6));
413 p->pitch_lag_int = pitch_lag_int; // store previous lag in a uint8_t
415 pitch_lag_frac <<= (p->cur_frame_mode != MODE_12k2);
417 pitch_lag_int += pitch_lag_frac > 0;
419 /* Calculate the pitch vector by interpolating the past excitation at the
420 pitch lag using a b60 hamming windowed sinc function. */
421 p->acelpf_ctx.acelp_interpolatef(p->excitation,
422 p->excitation + 1 - pitch_lag_int,
424 pitch_lag_frac + 6 - 6*(pitch_lag_frac > 0),
425 10, AMR_SUBFRAME_SIZE);
427 memcpy(p->pitch_vector, p->excitation, AMR_SUBFRAME_SIZE * sizeof(float));
433 /// @name AMR algebraic code book (fixed) vector decoding functions
437 * Decode a 10-bit algebraic codebook index from a 10.2 kbit/s frame.
439 static void decode_10bit_pulse(int code, int pulse_position[8],
440 int i1, int i2, int i3)
442 // coded using 7+3 bits with the 3 LSBs being, individually, the LSB of 1 of
443 // the 3 pulses and the upper 7 bits being coded in base 5
444 const uint8_t *positions = base_five_table[code >> 3];
445 pulse_position[i1] = (positions[2] << 1) + ( code & 1);
446 pulse_position[i2] = (positions[1] << 1) + ((code >> 1) & 1);
447 pulse_position[i3] = (positions[0] << 1) + ((code >> 2) & 1);
451 * Decode the algebraic codebook index to pulse positions and signs and
452 * construct the algebraic codebook vector for MODE_10k2.
454 * @param fixed_index positions of the eight pulses
455 * @param fixed_sparse pointer to the algebraic codebook vector
457 static void decode_8_pulses_31bits(const int16_t *fixed_index,
458 AMRFixed *fixed_sparse)
460 int pulse_position[8];
463 decode_10bit_pulse(fixed_index[4], pulse_position, 0, 4, 1);
464 decode_10bit_pulse(fixed_index[5], pulse_position, 2, 6, 5);
466 // coded using 5+2 bits with the 2 LSBs being, individually, the LSB of 1 of
467 // the 2 pulses and the upper 5 bits being coded in base 5
468 temp = ((fixed_index[6] >> 2) * 25 + 12) >> 5;
469 pulse_position[3] = temp % 5;
470 pulse_position[7] = temp / 5;
471 if (pulse_position[7] & 1)
472 pulse_position[3] = 4 - pulse_position[3];
473 pulse_position[3] = (pulse_position[3] << 1) + ( fixed_index[6] & 1);
474 pulse_position[7] = (pulse_position[7] << 1) + ((fixed_index[6] >> 1) & 1);
477 for (i = 0; i < 4; i++) {
478 const int pos1 = (pulse_position[i] << 2) + i;
479 const int pos2 = (pulse_position[i + 4] << 2) + i;
480 const float sign = fixed_index[i] ? -1.0 : 1.0;
481 fixed_sparse->x[i ] = pos1;
482 fixed_sparse->x[i + 4] = pos2;
483 fixed_sparse->y[i ] = sign;
484 fixed_sparse->y[i + 4] = pos2 < pos1 ? -sign : sign;
489 * Decode the algebraic codebook index to pulse positions and signs,
490 * then construct the algebraic codebook vector.
492 * nb of pulses | bits encoding pulses
493 * For MODE_4k75 or MODE_5k15, 2 | 1-3, 4-6, 7
494 * MODE_5k9, 2 | 1, 2-4, 5-6, 7-9
495 * MODE_6k7, 3 | 1-3, 4, 5-7, 8, 9-11
496 * MODE_7k4 or MODE_7k95, 4 | 1-3, 4-6, 7-9, 10, 11-13
498 * @param fixed_sparse pointer to the algebraic codebook vector
499 * @param pulses algebraic codebook indexes
500 * @param mode mode of the current frame
501 * @param subframe current subframe number
503 static void decode_fixed_sparse(AMRFixed *fixed_sparse, const uint16_t *pulses,
504 const enum Mode mode, const int subframe)
506 av_assert1(MODE_4k75 <= (signed)mode && mode <= MODE_12k2);
508 if (mode == MODE_12k2) {
509 ff_decode_10_pulses_35bits(pulses, fixed_sparse, gray_decode, 5, 3);
510 } else if (mode == MODE_10k2) {
511 decode_8_pulses_31bits(pulses, fixed_sparse);
513 int *pulse_position = fixed_sparse->x;
515 const int fixed_index = pulses[0];
517 if (mode <= MODE_5k15) {
518 pulse_subset = ((fixed_index >> 3) & 8) + (subframe << 1);
519 pulse_position[0] = ( fixed_index & 7) * 5 + track_position[pulse_subset];
520 pulse_position[1] = ((fixed_index >> 3) & 7) * 5 + track_position[pulse_subset + 1];
522 } else if (mode == MODE_5k9) {
523 pulse_subset = ((fixed_index & 1) << 1) + 1;
524 pulse_position[0] = ((fixed_index >> 1) & 7) * 5 + pulse_subset;
525 pulse_subset = (fixed_index >> 4) & 3;
526 pulse_position[1] = ((fixed_index >> 6) & 7) * 5 + pulse_subset + (pulse_subset == 3 ? 1 : 0);
527 fixed_sparse->n = pulse_position[0] == pulse_position[1] ? 1 : 2;
528 } else if (mode == MODE_6k7) {
529 pulse_position[0] = (fixed_index & 7) * 5;
530 pulse_subset = (fixed_index >> 2) & 2;
531 pulse_position[1] = ((fixed_index >> 4) & 7) * 5 + pulse_subset + 1;
532 pulse_subset = (fixed_index >> 6) & 2;
533 pulse_position[2] = ((fixed_index >> 8) & 7) * 5 + pulse_subset + 2;
535 } else { // mode <= MODE_7k95
536 pulse_position[0] = gray_decode[ fixed_index & 7];
537 pulse_position[1] = gray_decode[(fixed_index >> 3) & 7] + 1;
538 pulse_position[2] = gray_decode[(fixed_index >> 6) & 7] + 2;
539 pulse_subset = (fixed_index >> 9) & 1;
540 pulse_position[3] = gray_decode[(fixed_index >> 10) & 7] + pulse_subset + 3;
543 for (i = 0; i < fixed_sparse->n; i++)
544 fixed_sparse->y[i] = (pulses[1] >> i) & 1 ? 1.0 : -1.0;
549 * Apply pitch lag to obtain the sharpened fixed vector (section 6.1.2)
551 * @param p the context
552 * @param subframe unpacked amr subframe
553 * @param mode mode of the current frame
554 * @param fixed_sparse sparse respresentation of the fixed vector
556 static void pitch_sharpening(AMRContext *p, int subframe, enum Mode mode,
557 AMRFixed *fixed_sparse)
559 // The spec suggests the current pitch gain is always used, but in other
560 // modes the pitch and codebook gains are joinly quantized (sec 5.8.2)
561 // so the codebook gain cannot depend on the quantized pitch gain.
562 if (mode == MODE_12k2)
563 p->beta = FFMIN(p->pitch_gain[4], 1.0);
565 fixed_sparse->pitch_lag = p->pitch_lag_int;
566 fixed_sparse->pitch_fac = p->beta;
568 // Save pitch sharpening factor for the next subframe
569 // MODE_4k75 only updates on the 2nd and 4th subframes - this follows from
570 // the fact that the gains for two subframes are jointly quantized.
571 if (mode != MODE_4k75 || subframe & 1)
572 p->beta = av_clipf(p->pitch_gain[4], 0.0, SHARP_MAX);
577 /// @name AMR gain decoding functions
581 * fixed gain smoothing
582 * Note that where the spec specifies the "spectrum in the q domain"
583 * in section 6.1.4, in fact frequencies should be used.
585 * @param p the context
586 * @param lsf LSFs for the current subframe, in the range [0,1]
587 * @param lsf_avg averaged LSFs
588 * @param mode mode of the current frame
590 * @return fixed gain smoothed
592 static float fixed_gain_smooth(AMRContext *p , const float *lsf,
593 const float *lsf_avg, const enum Mode mode)
598 for (i = 0; i < LP_FILTER_ORDER; i++)
599 diff += fabs(lsf_avg[i] - lsf[i]) / lsf_avg[i];
601 // If diff is large for ten subframes, disable smoothing for a 40-subframe
607 if (p->diff_count > 10) {
609 p->diff_count--; // don't let diff_count overflow
612 if (p->hang_count < 40) {
614 } else if (mode < MODE_7k4 || mode == MODE_10k2) {
615 const float smoothing_factor = av_clipf(4.0 * diff - 1.6, 0.0, 1.0);
616 const float fixed_gain_mean = (p->fixed_gain[0] + p->fixed_gain[1] +
617 p->fixed_gain[2] + p->fixed_gain[3] +
618 p->fixed_gain[4]) * 0.2;
619 return smoothing_factor * p->fixed_gain[4] +
620 (1.0 - smoothing_factor) * fixed_gain_mean;
622 return p->fixed_gain[4];
626 * Decode pitch gain and fixed gain factor (part of section 6.1.3).
628 * @param p the context
629 * @param amr_subframe unpacked amr subframe
630 * @param mode mode of the current frame
631 * @param subframe current subframe number
632 * @param fixed_gain_factor decoded gain correction factor
634 static void decode_gains(AMRContext *p, const AMRNBSubframe *amr_subframe,
635 const enum Mode mode, const int subframe,
636 float *fixed_gain_factor)
638 if (mode == MODE_12k2 || mode == MODE_7k95) {
639 p->pitch_gain[4] = qua_gain_pit [amr_subframe->p_gain ]
641 *fixed_gain_factor = qua_gain_code[amr_subframe->fixed_gain]
644 const uint16_t *gains;
646 if (mode >= MODE_6k7) {
647 gains = gains_high[amr_subframe->p_gain];
648 } else if (mode >= MODE_5k15) {
649 gains = gains_low [amr_subframe->p_gain];
651 // gain index is only coded in subframes 0,2 for MODE_4k75
652 gains = gains_MODE_4k75[(p->frame.subframe[subframe & 2].p_gain << 1) + (subframe & 1)];
655 p->pitch_gain[4] = gains[0] * (1.0 / 16384.0);
656 *fixed_gain_factor = gains[1] * (1.0 / 4096.0);
663 /// @name AMR preprocessing functions
667 * Circularly convolve a sparse fixed vector with a phase dispersion impulse
668 * response filter (D.6.2 of G.729 and 6.1.5 of AMR).
670 * @param out vector with filter applied
671 * @param in source vector
672 * @param filter phase filter coefficients
674 * out[n] = sum(i,0,len-1){ in[i] * filter[(len + n - i)%len] }
676 static void apply_ir_filter(float *out, const AMRFixed *in,
679 float filter1[AMR_SUBFRAME_SIZE], ///< filters at pitch lag*1 and *2
680 filter2[AMR_SUBFRAME_SIZE];
681 int lag = in->pitch_lag;
682 float fac = in->pitch_fac;
685 if (lag < AMR_SUBFRAME_SIZE) {
686 ff_celp_circ_addf(filter1, filter, filter, lag, fac,
689 if (lag < AMR_SUBFRAME_SIZE >> 1)
690 ff_celp_circ_addf(filter2, filter, filter1, lag, fac,
694 memset(out, 0, sizeof(float) * AMR_SUBFRAME_SIZE);
695 for (i = 0; i < in->n; i++) {
698 const float *filterp;
700 if (x >= AMR_SUBFRAME_SIZE - lag) {
702 } else if (x >= AMR_SUBFRAME_SIZE - (lag << 1)) {
707 ff_celp_circ_addf(out, out, filterp, x, y, AMR_SUBFRAME_SIZE);
712 * Reduce fixed vector sparseness by smoothing with one of three IR filters.
713 * Also know as "adaptive phase dispersion".
715 * This implements 3GPP TS 26.090 section 6.1(5).
717 * @param p the context
718 * @param fixed_sparse algebraic codebook vector
719 * @param fixed_vector unfiltered fixed vector
720 * @param fixed_gain smoothed gain
721 * @param out space for modified vector if necessary
723 static const float *anti_sparseness(AMRContext *p, AMRFixed *fixed_sparse,
724 const float *fixed_vector,
725 float fixed_gain, float *out)
729 if (p->pitch_gain[4] < 0.6) {
730 ir_filter_nr = 0; // strong filtering
731 } else if (p->pitch_gain[4] < 0.9) {
732 ir_filter_nr = 1; // medium filtering
734 ir_filter_nr = 2; // no filtering
737 if (fixed_gain > 2.0 * p->prev_sparse_fixed_gain) {
738 p->ir_filter_onset = 2;
739 } else if (p->ir_filter_onset)
740 p->ir_filter_onset--;
742 if (!p->ir_filter_onset) {
745 for (i = 0; i < 5; i++)
746 if (p->pitch_gain[i] < 0.6)
751 if (ir_filter_nr > p->prev_ir_filter_nr + 1)
753 } else if (ir_filter_nr < 2)
756 // Disable filtering for very low level of fixed_gain.
757 // Note this step is not specified in the technical description but is in
758 // the reference source in the function Ph_disp.
759 if (fixed_gain < 5.0)
762 if (p->cur_frame_mode != MODE_7k4 && p->cur_frame_mode < MODE_10k2
763 && ir_filter_nr < 2) {
764 apply_ir_filter(out, fixed_sparse,
765 (p->cur_frame_mode == MODE_7k95 ?
766 ir_filters_lookup_MODE_7k95 :
767 ir_filters_lookup)[ir_filter_nr]);
771 // update ir filter strength history
772 p->prev_ir_filter_nr = ir_filter_nr;
773 p->prev_sparse_fixed_gain = fixed_gain;
781 /// @name AMR synthesis functions
785 * Conduct 10th order linear predictive coding synthesis.
787 * @param p pointer to the AMRContext
788 * @param lpc pointer to the LPC coefficients
789 * @param fixed_gain fixed codebook gain for synthesis
790 * @param fixed_vector algebraic codebook vector
791 * @param samples pointer to the output speech samples
792 * @param overflow 16-bit overflow flag
794 static int synthesis(AMRContext *p, float *lpc,
795 float fixed_gain, const float *fixed_vector,
796 float *samples, uint8_t overflow)
799 float excitation[AMR_SUBFRAME_SIZE];
801 // if an overflow has been detected, the pitch vector is scaled down by a
804 for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
805 p->pitch_vector[i] *= 0.25;
807 p->acelpv_ctx.weighted_vector_sumf(excitation, p->pitch_vector, fixed_vector,
808 p->pitch_gain[4], fixed_gain, AMR_SUBFRAME_SIZE);
810 // emphasize pitch vector contribution
811 if (p->pitch_gain[4] > 0.5 && !overflow) {
812 float energy = p->celpm_ctx.dot_productf(excitation, excitation,
816 (p->cur_frame_mode == MODE_12k2 ?
817 0.25 * FFMIN(p->pitch_gain[4], 1.0) :
818 0.5 * FFMIN(p->pitch_gain[4], SHARP_MAX));
820 for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
821 excitation[i] += pitch_factor * p->pitch_vector[i];
823 ff_scale_vector_to_given_sum_of_squares(excitation, excitation, energy,
827 p->celpf_ctx.celp_lp_synthesis_filterf(samples, lpc, excitation,
832 for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
833 if (fabsf(samples[i]) > AMR_SAMPLE_BOUND) {
843 /// @name 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 /// @name AMR Postprocessing functions
872 * Get the tilt factor of a formant filter from its transfer function
874 * @param p The Context
875 * @param lpc_n LP_FILTER_ORDER coefficients of the numerator
876 * @param lpc_d LP_FILTER_ORDER coefficients of the denominator
878 static float tilt_factor(AMRContext *p, float *lpc_n, float *lpc_d)
880 float rh0, rh1; // autocorrelation at lag 0 and 1
882 // LP_FILTER_ORDER prior zeros are needed for ff_celp_lp_synthesis_filterf
883 float impulse_buffer[LP_FILTER_ORDER + AMR_TILT_RESPONSE] = { 0 };
884 float *hf = impulse_buffer + LP_FILTER_ORDER; // start of impulse response
887 memcpy(hf + 1, lpc_n, sizeof(float) * LP_FILTER_ORDER);
888 p->celpf_ctx.celp_lp_synthesis_filterf(hf, lpc_d, hf,
892 rh0 = p->celpm_ctx.dot_productf(hf, hf, AMR_TILT_RESPONSE);
893 rh1 = p->celpm_ctx.dot_productf(hf, hf + 1, AMR_TILT_RESPONSE - 1);
895 // The spec only specifies this check for 12.2 and 10.2 kbit/s
896 // modes. But in the ref source the tilt is always non-negative.
897 return rh1 >= 0.0 ? rh1 / rh0 * AMR_TILT_GAMMA_T : 0.0;
901 * Perform adaptive post-filtering to enhance the quality of the speech.
904 * @param p pointer to the AMRContext
905 * @param lpc interpolated LP coefficients for this subframe
906 * @param buf_out output of the filter
908 static void postfilter(AMRContext *p, float *lpc, float *buf_out)
911 float *samples = p->samples_in + LP_FILTER_ORDER; // Start of input
913 float speech_gain = p->celpm_ctx.dot_productf(samples, samples,
916 float pole_out[AMR_SUBFRAME_SIZE + LP_FILTER_ORDER]; // Output of pole filter
917 const float *gamma_n, *gamma_d; // Formant filter factor table
918 float lpc_n[LP_FILTER_ORDER], lpc_d[LP_FILTER_ORDER]; // Transfer function coefficients
920 if (p->cur_frame_mode == MODE_12k2 || p->cur_frame_mode == MODE_10k2) {
921 gamma_n = ff_pow_0_7;
922 gamma_d = ff_pow_0_75;
924 gamma_n = ff_pow_0_55;
925 gamma_d = ff_pow_0_7;
928 for (i = 0; i < LP_FILTER_ORDER; i++) {
929 lpc_n[i] = lpc[i] * gamma_n[i];
930 lpc_d[i] = lpc[i] * gamma_d[i];
933 memcpy(pole_out, p->postfilter_mem, sizeof(float) * LP_FILTER_ORDER);
934 p->celpf_ctx.celp_lp_synthesis_filterf(pole_out + LP_FILTER_ORDER, lpc_d, samples,
935 AMR_SUBFRAME_SIZE, LP_FILTER_ORDER);
936 memcpy(p->postfilter_mem, pole_out + AMR_SUBFRAME_SIZE,
937 sizeof(float) * LP_FILTER_ORDER);
939 p->celpf_ctx.celp_lp_zero_synthesis_filterf(buf_out, lpc_n,
940 pole_out + LP_FILTER_ORDER,
941 AMR_SUBFRAME_SIZE, LP_FILTER_ORDER);
943 ff_tilt_compensation(&p->tilt_mem, tilt_factor(p, lpc_n, lpc_d), buf_out,
946 ff_adaptive_gain_control(buf_out, buf_out, speech_gain, AMR_SUBFRAME_SIZE,
947 AMR_AGC_ALPHA, &p->postfilter_agc);
952 static int amrnb_decode_frame(AVCodecContext *avctx, void *data,
953 int *got_frame_ptr, AVPacket *avpkt)
956 AMRContext *p = avctx->priv_data; // pointer to private data
957 const uint8_t *buf = avpkt->data;
958 int buf_size = avpkt->size;
959 float *buf_out; // pointer to the output data buffer
960 int i, subframe, ret;
961 float fixed_gain_factor;
962 AMRFixed fixed_sparse = {0}; // fixed vector up to anti-sparseness processing
963 float spare_vector[AMR_SUBFRAME_SIZE]; // extra stack space to hold result from anti-sparseness processing
964 float synth_fixed_gain; // the fixed gain that synthesis should use
965 const float *synth_fixed_vector; // pointer to the fixed vector that synthesis should use
967 /* get output buffer */
968 p->avframe.nb_samples = AMR_BLOCK_SIZE;
969 if ((ret = ff_get_buffer(avctx, &p->avframe)) < 0) {
970 av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
973 buf_out = (float *)p->avframe.data[0];
975 p->cur_frame_mode = unpack_bitstream(p, buf, buf_size);
976 if (p->cur_frame_mode == NO_DATA) {
977 av_log(avctx, AV_LOG_ERROR, "Corrupt bitstream\n");
978 return AVERROR_INVALIDDATA;
980 if (p->cur_frame_mode == MODE_DTX) {
981 av_log_missing_feature(avctx, "dtx mode", 0);
982 av_log(avctx, AV_LOG_INFO, "Note: libopencore_amrnb supports dtx\n");
983 return AVERROR_PATCHWELCOME;
986 if (p->cur_frame_mode == MODE_12k2) {
991 for (i = 0; i < 4; i++)
992 ff_acelp_lspd2lpc(p->lsp[i], p->lpc[i], 5);
994 for (subframe = 0; subframe < 4; subframe++) {
995 const AMRNBSubframe *amr_subframe = &p->frame.subframe[subframe];
997 decode_pitch_vector(p, amr_subframe, subframe);
999 decode_fixed_sparse(&fixed_sparse, amr_subframe->pulses,
1000 p->cur_frame_mode, subframe);
1002 // The fixed gain (section 6.1.3) depends on the fixed vector
1003 // (section 6.1.2), but the fixed vector calculation uses
1004 // pitch sharpening based on the on the pitch gain (section 6.1.3).
1005 // So the correct order is: pitch gain, pitch sharpening, fixed gain.
1006 decode_gains(p, amr_subframe, p->cur_frame_mode, subframe,
1007 &fixed_gain_factor);
1009 pitch_sharpening(p, subframe, p->cur_frame_mode, &fixed_sparse);
1011 if (fixed_sparse.pitch_lag == 0) {
1012 av_log(avctx, AV_LOG_ERROR, "The file is corrupted, pitch_lag = 0 is not allowed\n");
1013 return AVERROR_INVALIDDATA;
1015 ff_set_fixed_vector(p->fixed_vector, &fixed_sparse, 1.0,
1019 ff_amr_set_fixed_gain(fixed_gain_factor,
1020 p->celpm_ctx.dot_productf(p->fixed_vector,
1022 AMR_SUBFRAME_SIZE) /
1024 p->prediction_error,
1025 energy_mean[p->cur_frame_mode], energy_pred_fac);
1027 // The excitation feedback is calculated without any processing such
1028 // as fixed gain smoothing. This isn't mentioned in the specification.
1029 for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
1030 p->excitation[i] *= p->pitch_gain[4];
1031 ff_set_fixed_vector(p->excitation, &fixed_sparse, p->fixed_gain[4],
1034 // In the ref decoder, excitation is stored with no fractional bits.
1035 // This step prevents buzz in silent periods. The ref encoder can
1036 // emit long sequences with pitch factor greater than one. This
1037 // creates unwanted feedback if the excitation vector is nonzero.
1038 // (e.g. test sequence T19_795.COD in 3GPP TS 26.074)
1039 for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
1040 p->excitation[i] = truncf(p->excitation[i]);
1042 // Smooth fixed gain.
1043 // The specification is ambiguous, but in the reference source, the
1044 // smoothed value is NOT fed back into later fixed gain smoothing.
1045 synth_fixed_gain = fixed_gain_smooth(p, p->lsf_q[subframe],
1046 p->lsf_avg, p->cur_frame_mode);
1048 synth_fixed_vector = anti_sparseness(p, &fixed_sparse, p->fixed_vector,
1049 synth_fixed_gain, spare_vector);
1051 if (synthesis(p, p->lpc[subframe], synth_fixed_gain,
1052 synth_fixed_vector, &p->samples_in[LP_FILTER_ORDER], 0))
1053 // overflow detected -> rerun synthesis scaling pitch vector down
1054 // by a factor of 4, skipping pitch vector contribution emphasis
1055 // and adaptive gain control
1056 synthesis(p, p->lpc[subframe], synth_fixed_gain,
1057 synth_fixed_vector, &p->samples_in[LP_FILTER_ORDER], 1);
1059 postfilter(p, p->lpc[subframe], buf_out + subframe * AMR_SUBFRAME_SIZE);
1061 // update buffers and history
1062 ff_clear_fixed_vector(p->fixed_vector, &fixed_sparse, AMR_SUBFRAME_SIZE);
1066 p->acelpf_ctx.acelp_apply_order_2_transfer_function(buf_out,
1067 buf_out, highpass_zeros,
1069 highpass_gain * AMR_SAMPLE_SCALE,
1070 p->high_pass_mem, AMR_BLOCK_SIZE);
1072 /* Update averaged lsf vector (used for fixed gain smoothing).
1074 * Note that lsf_avg should not incorporate the current frame's LSFs
1075 * for fixed_gain_smooth.
1076 * The specification has an incorrect formula: the reference decoder uses
1077 * qbar(n-1) rather than qbar(n) in section 6.1(4) equation 71. */
1078 p->acelpv_ctx.weighted_vector_sumf(p->lsf_avg, p->lsf_avg, p->lsf_q[3],
1079 0.84, 0.16, LP_FILTER_ORDER);
1082 *(AVFrame *)data = p->avframe;
1084 /* return the amount of bytes consumed if everything was OK */
1085 return frame_sizes_nb[p->cur_frame_mode] + 1; // +7 for rounding and +8 for TOC
1089 AVCodec ff_amrnb_decoder = {
1091 .type = AVMEDIA_TYPE_AUDIO,
1092 .id = AV_CODEC_ID_AMR_NB,
1093 .priv_data_size = sizeof(AMRContext),
1094 .init = amrnb_decode_init,
1095 .decode = amrnb_decode_frame,
1096 .capabilities = CODEC_CAP_DR1,
1097 .long_name = NULL_IF_CONFIG_SMALL("AMR-NB (Adaptive Multi-Rate NarrowBand)"),
1098 .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT,
1099 AV_SAMPLE_FMT_NONE },