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acelp_pitch_delay.c
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1 /*
2  * gain code, gain pitch and pitch delay decoding
3  *
4  * Copyright (c) 2008 Vladimir Voroshilov
5  *
6  * This file is part of FFmpeg.
7  *
8  * FFmpeg is free software; you can redistribute it and/or
9  * modify it under the terms of the GNU Lesser General Public
10  * License as published by the Free Software Foundation; either
11  * version 2.1 of the License, or (at your option) any later version.
12  *
13  * FFmpeg is distributed in the hope that it will be useful,
14  * but WITHOUT ANY WARRANTY; without even the implied warranty of
15  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
16  * Lesser General Public License for more details.
17  *
18  * You should have received a copy of the GNU Lesser General Public
19  * License along with FFmpeg; if not, write to the Free Software
20  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
21  */
22 
23 #include "libavutil/common.h"
24 #include "libavutil/ffmath.h"
25 #include "libavutil/float_dsp.h"
26 #include "libavutil/mathematics.h"
27 #include "avcodec.h"
28 #include "acelp_pitch_delay.h"
29 #include "celp_math.h"
30 #include "audiodsp.h"
31 
33 {
34  ac_index += 58;
35  if(ac_index > 254)
36  ac_index = 3 * ac_index - 510;
37  return ac_index;
38 }
39 
41  int ac_index,
42  int pitch_delay_min)
43 {
44  if(ac_index < 4)
45  return 3 * (ac_index + pitch_delay_min);
46  else if(ac_index < 12)
47  return 3 * pitch_delay_min + ac_index + 6;
48  else
49  return 3 * (ac_index + pitch_delay_min) - 18;
50 }
51 
53  int ac_index,
54  int pitch_delay_min)
55 {
56  return 3 * pitch_delay_min + ac_index - 2;
57 }
58 
60 {
61  if(ac_index < 463)
62  return ac_index + 105;
63  else
64  return 6 * (ac_index - 368);
65 }
67  int ac_index,
68  int pitch_delay_min)
69 {
70  return 6 * pitch_delay_min + ac_index - 3;
71 }
72 
74  int16_t* quant_energy,
75  int gain_corr_factor,
76  int log2_ma_pred_order,
77  int erasure)
78 {
79  int i;
80  int avg_gain=quant_energy[(1 << log2_ma_pred_order) - 1]; // (5.10)
81 
82  for(i=(1 << log2_ma_pred_order) - 1; i>0; i--)
83  {
84  avg_gain += quant_energy[i-1];
85  quant_energy[i] = quant_energy[i-1];
86  }
87 
88  if(erasure)
89  quant_energy[0] = FFMAX(avg_gain >> log2_ma_pred_order, -10240) - 4096; // -10 and -4 in (5.10)
90  else
91  quant_energy[0] = (6165 * ((ff_log2_q15(gain_corr_factor) >> 2) - (13 << 13))) >> 13;
92 }
93 
95  AudioDSPContext *adsp,
96  int gain_corr_factor,
97  const int16_t* fc_v,
98  int mr_energy,
99  const int16_t* quant_energy,
100  const int16_t* ma_prediction_coeff,
101  int subframe_size,
102  int ma_pred_order)
103 {
104  int i;
105 
106  mr_energy <<= 10;
107 
108  for(i=0; i<ma_pred_order; i++)
109  mr_energy += quant_energy[i] * ma_prediction_coeff[i];
110 
111 #ifdef G729_BITEXACT
112  mr_energy += (((-6165LL * ff_log2(dsp->scalarproduct_int16(fc_v, fc_v, subframe_size, 0))) >> 3) & ~0x3ff);
113 
114  mr_energy = (5439 * (mr_energy >> 15)) >> 8; // (0.15) = (0.15) * (7.23)
115 
116  return bidir_sal(
117  ((ff_exp2(mr_energy & 0x7fff) + 16) >> 5) * (gain_corr_factor >> 1),
118  (mr_energy >> 15) - 25
119  );
120 #else
121  mr_energy = gain_corr_factor * exp(M_LN10 / (20 << 23) * mr_energy) /
122  sqrt(adsp->scalarproduct_int16(fc_v, fc_v, subframe_size));
123  return mr_energy >> 12;
124 #endif
125 }
126 
127 float ff_amr_set_fixed_gain(float fixed_gain_factor, float fixed_mean_energy,
128  float *prediction_error, float energy_mean,
129  const float *pred_table)
130 {
131  // Equations 66-69:
132  // ^g_c = ^gamma_gc * 100.05 (predicted dB + mean dB - dB of fixed vector)
133  // Note 10^(0.05 * -10log(average x2)) = 1/sqrt((average x2)).
134  float val = fixed_gain_factor *
135  ff_exp10(0.05 *
136  (avpriv_scalarproduct_float_c(pred_table, prediction_error, 4) +
137  energy_mean)) /
138  sqrtf(fixed_mean_energy);
139 
140  // update quantified prediction error energy history
141  memmove(&prediction_error[0], &prediction_error[1],
142  3 * sizeof(prediction_error[0]));
143  prediction_error[3] = 20.0 * log10f(fixed_gain_factor);
144 
145  return val;
146 }
147 
148 void ff_decode_pitch_lag(int *lag_int, int *lag_frac, int pitch_index,
149  const int prev_lag_int, const int subframe,
150  int third_as_first, int resolution)
151 {
152  /* Note n * 10923 >> 15 is floor(x/3) for 0 <= n <= 32767 */
153  if (subframe == 0 || (subframe == 2 && third_as_first)) {
154 
155  if (pitch_index < 197)
156  pitch_index += 59;
157  else
158  pitch_index = 3 * pitch_index - 335;
159 
160  } else {
161  if (resolution == 4) {
162  int search_range_min = av_clip(prev_lag_int - 5, PITCH_DELAY_MIN,
163  PITCH_DELAY_MAX - 9);
164 
165  // decoding with 4-bit resolution
166  if (pitch_index < 4) {
167  // integer only precision for [search_range_min, search_range_min+3]
168  pitch_index = 3 * (pitch_index + search_range_min) + 1;
169  } else if (pitch_index < 12) {
170  // 1/3 fractional precision for [search_range_min+3 1/3, search_range_min+5 2/3]
171  pitch_index += 3 * search_range_min + 7;
172  } else {
173  // integer only precision for [search_range_min+6, search_range_min+9]
174  pitch_index = 3 * (pitch_index + search_range_min - 6) + 1;
175  }
176  } else {
177  // decoding with 5 or 6 bit resolution, 1/3 fractional precision
178  pitch_index--;
179 
180  if (resolution == 5) {
181  pitch_index += 3 * av_clip(prev_lag_int - 10, PITCH_DELAY_MIN,
182  PITCH_DELAY_MAX - 19);
183  } else
184  pitch_index += 3 * av_clip(prev_lag_int - 5, PITCH_DELAY_MIN,
185  PITCH_DELAY_MAX - 9);
186  }
187  }
188  *lag_int = pitch_index * 10923 >> 15;
189  *lag_frac = pitch_index - 3 * *lag_int - 1;
190 }
void ff_decode_pitch_lag(int *lag_int, int *lag_frac, int pitch_index, const int prev_lag_int, const int subframe, int third_as_first, int resolution)
Decode the adaptive codebook index to the integer and fractional parts of the pitch lag for one subfr...
const char const char void * val
Definition: avisynth_c.h:771
int32_t(* scalarproduct_int16)(const int16_t *v1, const int16_t *v2, int len)
Calculate scalar product of two vectors.
Definition: audiodsp.h:29
static const uint16_t ma_prediction_coeff[4]
MA prediction coefficients (3.9.1 of G.729, near Equation 69)
Definition: g729data.h:343
int ff_exp2(uint16_t power)
fixed-point implementation of exp2(x) in [0; 1] domain.
Definition: celp_math.c:48
int ff_acelp_decode_6bit_to_2nd_delay6(int ac_index, int pitch_delay_min)
Decode pitch delay of the second subframe encoded by 6 bits with 1/6 precision.
int ff_acelp_decode_4bit_to_2nd_delay3(int ac_index, int pitch_delay_min)
Decode pitch delay with 1/3 precision.
int ff_acelp_decode_8bit_to_1st_delay3(int ac_index)
Decode pitch delay of the first subframe encoded by 8 bits with 1/3 resolution.
int ff_acelp_decode_9bit_to_1st_delay6(int ac_index)
Decode pitch delay of the first subframe encoded by 9 bits with 1/6 precision.
float avpriv_scalarproduct_float_c(const float *v1, const float *v2, int len)
Return the scalar product of two vectors.
Definition: float_dsp.c:116
#define PITCH_DELAY_MAX
int ff_acelp_decode_5_6_bit_to_2nd_delay3(int ac_index, int pitch_delay_min)
Decode pitch delay of the second subframe encoded by 5 or 6 bits with 1/3 precision.
static av_always_inline double ff_exp10(double x)
Compute 10^x for floating point values.
Definition: ffmath.h:42
int ff_log2_q15(uint32_t value)
Calculate log2(x).
Definition: celp_math.c:78
#define FFMAX(a, b)
Definition: common.h:94
int8_t exp
Definition: eval.c:65
void ff_acelp_update_past_gain(int16_t *quant_energy, int gain_corr_factor, int log2_ma_pred_order, int erasure)
Update past quantized energies.
Libavcodec external API header.
#define ff_log2
Definition: intmath.h:50
#define M_LN10
Definition: mathematics.h:43
internal math functions header
common internal and external API header
static int bidir_sal(int value, int offset)
Shift value left or right depending on sign of offset parameter.
Definition: celp_math.h:81
static const float energy_mean[8]
desired mean innovation energy, indexed by active mode
Definition: amrnbdata.h:1458
#define PITCH_DELAY_MIN
#define log10f(x)
Definition: libm.h:414
int16_t ff_acelp_decode_gain_code(AudioDSPContext *adsp, int gain_corr_factor, const int16_t *fc_v, int mr_energy, const int16_t *quant_energy, const int16_t *ma_prediction_coeff, int subframe_size, int ma_pred_order)
Decode the adaptive codebook gain and add correction (4.1.5 and 3.9.1 of G.729).
float ff_amr_set_fixed_gain(float fixed_gain_factor, float fixed_mean_energy, float *prediction_error, float energy_mean, const float *pred_table)
Calculate fixed gain (part of section 6.1.3 of AMR spec)