FFmpeg
acelp_pitch_delay.c
Go to the documentation of this file.
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  int16_t* quant_energy,
34  int gain_corr_factor,
35  int log2_ma_pred_order,
36  int erasure)
37 {
38  int i;
39  int avg_gain=quant_energy[(1 << log2_ma_pred_order) - 1]; // (5.10)
40 
41  for(i=(1 << log2_ma_pred_order) - 1; i>0; i--)
42  {
43  avg_gain += quant_energy[i-1];
44  quant_energy[i] = quant_energy[i-1];
45  }
46 
47  if(erasure)
48  quant_energy[0] = FFMAX(avg_gain >> log2_ma_pred_order, -10240) - 4096; // -10 and -4 in (5.10)
49  else
50  quant_energy[0] = (6165 * ((ff_log2_q15(gain_corr_factor) >> 2) - (13 << 13))) >> 13;
51 }
52 
54  AudioDSPContext *adsp,
55  int gain_corr_factor,
56  const int16_t* fc_v,
57  int mr_energy,
58  const int16_t* quant_energy,
59  const int16_t* ma_prediction_coeff,
60  int subframe_size,
61  int ma_pred_order)
62 {
63  int i;
64 
65  mr_energy <<= 10;
66 
67  for(i=0; i<ma_pred_order; i++)
68  mr_energy += quant_energy[i] * ma_prediction_coeff[i];
69 
70 #ifdef G729_BITEXACT
71  mr_energy += (((-6165LL * ff_log2(dsp->scalarproduct_int16(fc_v, fc_v, subframe_size, 0))) >> 3) & ~0x3ff);
72 
73  mr_energy = (5439 * (mr_energy >> 15)) >> 8; // (0.15) = (0.15) * (7.23)
74 
75  return bidir_sal(
76  ((ff_exp2(mr_energy & 0x7fff) + 16) >> 5) * (gain_corr_factor >> 1),
77  (mr_energy >> 15) - 25
78  );
79 #else
80  mr_energy = gain_corr_factor * ff_exp10((double)mr_energy / (20 << 23)) /
81  sqrt(adsp->scalarproduct_int16(fc_v, fc_v, subframe_size));
82  return mr_energy >> 12;
83 #endif
84 }
85 
86 float ff_amr_set_fixed_gain(float fixed_gain_factor, float fixed_mean_energy,
87  float *prediction_error, float energy_mean,
88  const float *pred_table)
89 {
90  // Equations 66-69:
91  // ^g_c = ^gamma_gc * 100.05 (predicted dB + mean dB - dB of fixed vector)
92  // Note 10^(0.05 * -10log(average x2)) = 1/sqrt((average x2)).
93  float val = fixed_gain_factor *
94  ff_exp10(0.05 *
95  (avpriv_scalarproduct_float_c(pred_table, prediction_error, 4) +
96  energy_mean)) /
97  sqrtf(fixed_mean_energy ? fixed_mean_energy : 1.0);
98 
99  // update quantified prediction error energy history
100  memmove(&prediction_error[0], &prediction_error[1],
101  3 * sizeof(prediction_error[0]));
102  prediction_error[3] = 20.0 * log10f(fixed_gain_factor);
103 
104  return val;
105 }
106 
107 void ff_decode_pitch_lag(int *lag_int, int *lag_frac, int pitch_index,
108  const int prev_lag_int, const int subframe,
109  int third_as_first, int resolution)
110 {
111  /* Note n * 10923 >> 15 is floor(x/3) for 0 <= n <= 32767 */
112  if (subframe == 0 || (subframe == 2 && third_as_first)) {
113 
114  if (pitch_index < 197)
115  pitch_index += 59;
116  else
117  pitch_index = 3 * pitch_index - 335;
118 
119  } else {
120  if (resolution == 4) {
121  int search_range_min = av_clip(prev_lag_int - 5, PITCH_DELAY_MIN,
122  PITCH_DELAY_MAX - 9);
123 
124  // decoding with 4-bit resolution
125  if (pitch_index < 4) {
126  // integer only precision for [search_range_min, search_range_min+3]
127  pitch_index = 3 * (pitch_index + search_range_min) + 1;
128  } else if (pitch_index < 12) {
129  // 1/3 fractional precision for [search_range_min+3 1/3, search_range_min+5 2/3]
130  pitch_index += 3 * search_range_min + 7;
131  } else {
132  // integer only precision for [search_range_min+6, search_range_min+9]
133  pitch_index = 3 * (pitch_index + search_range_min - 6) + 1;
134  }
135  } else {
136  // decoding with 5 or 6 bit resolution, 1/3 fractional precision
137  pitch_index--;
138 
139  if (resolution == 5) {
140  pitch_index += 3 * av_clip(prev_lag_int - 10, PITCH_DELAY_MIN,
141  PITCH_DELAY_MAX - 19);
142  } else
143  pitch_index += 3 * av_clip(prev_lag_int - 5, PITCH_DELAY_MIN,
144  PITCH_DELAY_MAX - 9);
145  }
146  }
147  *lag_int = pitch_index * 10923 >> 15;
148  *lag_frac = pitch_index - 3 * *lag_int - 1;
149 }
ff_exp10
static av_always_inline double ff_exp10(double x)
Compute 10^x for floating point values.
Definition: ffmath.h:42
av_clip
#define av_clip
Definition: common.h:96
ff_amr_set_fixed_gain
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)
Definition: acelp_pitch_delay.c:86
ff_decode_pitch_lag
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...
Definition: acelp_pitch_delay.c:107
mathematics.h
FFMAX
#define FFMAX(a, b)
Definition: macros.h:47
PITCH_DELAY_MAX
#define PITCH_DELAY_MAX
Definition: acelp_pitch_delay.h:31
resolution
The official guide to swscale for confused that consecutive non overlapping rectangles of slice_bottom special converter These generally are unscaled converters of common like for each output line the vertical scaler pulls lines from a ring buffer When the ring buffer does not contain the wanted then it is pulled from the input slice through the input converter and horizontal scaler The result is also stored in the ring buffer to serve future vertical scaler requests When no more output can be generated because lines from a future slice would be then all remaining lines in the current slice are horizontally scaled and put in the ring buffer[This is done for luma and chroma, each with possibly different numbers of lines per picture.] Input to YUV Converter When the input to the main path is not planar bits per component YUV or bit it is converted to planar bit YUV Two sets of converters exist for this the other leaves the full chroma resolution
Definition: swscale.txt:54
log10f
#define log10f(x)
Definition: libm.h:414
val
static double val(void *priv, double ch)
Definition: aeval.c:76
ff_log2_q15
int ff_log2_q15(uint32_t value)
Calculate log2(x).
Definition: celp_math.c:80
ff_log2
#define ff_log2
Definition: intmath.h:50
ff_acelp_update_past_gain
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.
Definition: acelp_pitch_delay.c:32
float_dsp.h
ma_prediction_coeff
static const uint16_t ma_prediction_coeff[4]
MA prediction coefficients (3.9.1 of G.729, near Equation 69)
Definition: g729data.h:343
celp_math.h
AudioDSPContext::scalarproduct_int16
int32_t(* scalarproduct_int16)(const int16_t *v1, const int16_t *v2, int len)
Calculate scalar product of two vectors.
Definition: audiodsp.h:29
i
int i
Definition: input.c:406
common.h
avcodec.h
energy_mean
static const float energy_mean[8]
desired mean innovation energy, indexed by active mode
Definition: amrnbdata.h:1453
ffmath.h
avpriv_scalarproduct_float_c
float avpriv_scalarproduct_float_c(const float *v1, const float *v2, int len)
Return the scalar product of two vectors.
Definition: float_dsp.c:124
audiodsp.h
bidir_sal
static int bidir_sal(int value, int offset)
Shift value left or right depending on sign of offset parameter.
Definition: celp_math.h:81
ff_exp2
int ff_exp2(uint16_t power)
fixed-point implementation of exp2(x) in [0; 1] domain.
AudioDSPContext
Definition: audiodsp.h:24
ff_acelp_decode_gain_code
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).
Definition: acelp_pitch_delay.c:53
PITCH_DELAY_MIN
#define PITCH_DELAY_MIN
Definition: acelp_pitch_delay.h:30
acelp_pitch_delay.h