11235
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1 /*
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2 * AMR narrowband decoder
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3 * Copyright (c) 2006-2007 Robert Swain
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4 * Copyright (c) 2009 Colin McQuillan
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5 *
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6 * This file is part of FFmpeg.
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7 *
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8 * FFmpeg is free software; you can redistribute it and/or
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9 * modify it under the terms of the GNU Lesser General Public
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10 * License as published by the Free Software Foundation; either
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11 * version 2.1 of the License, or (at your option) any later version.
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12 *
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13 * FFmpeg is distributed in the hope that it will be useful,
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14 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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16 * Lesser General Public License for more details.
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17 *
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18 * You should have received a copy of the GNU Lesser General Public
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19 * License along with FFmpeg; if not, write to the Free Software
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20 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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21 */
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22
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23
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24 /**
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25 * @file libavcodec/amrnbdec.c
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26 * AMR narrowband decoder
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27 *
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28 * This decoder uses floats for simplicity and so is not bit-exact. One
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29 * difference is that differences in phase can accumulate. The test sequences
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30 * in 3GPP TS 26.074 can still be useful.
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31 *
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32 * - Comparing this file's output to the output of the ref decoder gives a
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33 * PSNR of 30 to 80. Plotting the output samples shows a difference in
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34 * phase in some areas.
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35 *
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36 * - Comparing both decoders against their input, this decoder gives a similar
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37 * PSNR. If the test sequence homing frames are removed (this decoder does
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38 * not detect them), the PSNR is at least as good as the reference on 140
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39 * out of 169 tests.
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40 */
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41
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42
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43 #include <string.h>
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44 #include <math.h>
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45
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46 #include "avcodec.h"
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47 #include "get_bits.h"
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48 #include "libavutil/common.h"
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49 #include "celp_math.h"
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50 #include "celp_filters.h"
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51 #include "acelp_filters.h"
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52 #include "acelp_vectors.h"
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53 #include "acelp_pitch_delay.h"
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54 #include "lsp.h"
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55
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56 #include "amrnbdata.h"
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57
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58 #define AMR_BLOCK_SIZE 160 ///< samples per frame
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59 #define AMR_SAMPLE_BOUND 32768.0 ///< threshold for synthesis overflow
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60
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61 /**
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62 * Scale from constructed speech to [-1,1]
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63 *
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64 * AMR is designed to produce 16-bit PCM samples (3GPP TS 26.090 4.2) but
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65 * upscales by two (section 6.2.2).
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66 *
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67 * Fundamentally, this scale is determined by energy_mean through
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68 * the fixed vector contribution to the excitation vector.
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69 */
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70 #define AMR_SAMPLE_SCALE (2.0 / 32768.0)
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71
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72 /** Prediction factor for 12.2kbit/s mode */
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73 #define PRED_FAC_MODE_12k2 0.65
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74
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75 #define LSF_R_FAC (8000.0 / 32768.0) ///< LSF residual tables to Hertz
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76 #define MIN_LSF_SPACING (50.0488 / 8000.0) ///< Ensures stability of LPC filter
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77 #define PITCH_LAG_MIN_MODE_12k2 18 ///< Lower bound on decoded lag search in 12.2kbit/s mode
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78
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79 /** Initial energy in dB. Also used for bad frames (unimplemented). */
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80 #define MIN_ENERGY -14.0
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81
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82 /** Maximum sharpening factor
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83 *
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84 * The specification says 0.8, which should be 13107, but the reference C code
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85 * uses 13017 instead. (Amusingly the same applies to SHARP_MAX in g729dec.c.)
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86 */
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87 #define SHARP_MAX 0.79449462890625
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88
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89 /** Number of impulse response coefficients used for tilt factor */
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90 #define AMR_TILT_RESPONSE 22
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91 /** Tilt factor = 1st reflection coefficient * gamma_t */
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92 #define AMR_TILT_GAMMA_T 0.8
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93 /** Adaptive gain control factor used in post-filter */
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94 #define AMR_AGC_ALPHA 0.9
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95
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96 typedef struct AMRContext {
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97 AMRNBFrame frame; ///< decoded AMR parameters (lsf coefficients, codebook indexes, etc)
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98 uint8_t bad_frame_indicator; ///< bad frame ? 1 : 0
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99 enum Mode cur_frame_mode;
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100
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101 int16_t prev_lsf_r[LP_FILTER_ORDER]; ///< residual LSF vector from previous subframe
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102 double lsp[4][LP_FILTER_ORDER]; ///< lsp vectors from current frame
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103 double prev_lsp_sub4[LP_FILTER_ORDER]; ///< lsp vector for the 4th subframe of the previous frame
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104
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105 float lsf_q[4][LP_FILTER_ORDER]; ///< Interpolated LSF vector for fixed gain smoothing
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106 float lsf_avg[LP_FILTER_ORDER]; ///< vector of averaged lsf vector
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107
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108 float lpc[4][LP_FILTER_ORDER]; ///< lpc coefficient vectors for 4 subframes
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109
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110 uint8_t pitch_lag_int; ///< integer part of pitch lag from current subframe
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111
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112 float excitation_buf[PITCH_DELAY_MAX + LP_FILTER_ORDER + 1 + AMR_SUBFRAME_SIZE]; ///< current excitation and all necessary excitation history
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113 float *excitation; ///< pointer to the current excitation vector in excitation_buf
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114
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115 float pitch_vector[AMR_SUBFRAME_SIZE]; ///< adaptive code book (pitch) vector
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116 float fixed_vector[AMR_SUBFRAME_SIZE]; ///< algebraic codebook (fixed) vector (must be kept zero between frames)
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117
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118 float prediction_error[4]; ///< quantified prediction errors {20log10(^gamma_gc)} for previous four subframes
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119 float pitch_gain[5]; ///< quantified pitch gains for the current and previous four subframes
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120 float fixed_gain[5]; ///< quantified fixed gains for the current and previous four subframes
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121
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122 float beta; ///< previous pitch_gain, bounded by [0.0,SHARP_MAX]
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123 uint8_t diff_count; ///< the number of subframes for which diff has been above 0.65
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124 uint8_t hang_count; ///< the number of subframes since a hangover period started
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125
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126 float prev_sparse_fixed_gain; ///< previous fixed gain; used by anti-sparseness processing to determine "onset"
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127 uint8_t prev_ir_filter_nr; ///< previous impulse response filter "impNr": 0 - strong, 1 - medium, 2 - none
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128 uint8_t ir_filter_onset; ///< flag for impulse response filter strength
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129
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130 float postfilter_mem[10]; ///< previous intermediate values in the formant filter
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131 float tilt_mem; ///< previous input to tilt compensation filter
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132 float postfilter_agc; ///< previous factor used for adaptive gain control
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133 float high_pass_mem[2]; ///< previous intermediate values in the high-pass filter
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134
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135 float samples_in[LP_FILTER_ORDER + AMR_SUBFRAME_SIZE]; ///< floating point samples
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136
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137 } AMRContext;
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138
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139 /** Double version of ff_weighted_vector_sumf() */
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140 static void weighted_vector_sumd(double *out, const double *in_a,
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141 const double *in_b, double weight_coeff_a,
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142 double weight_coeff_b, int length)
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143 {
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144 int i;
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145
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146 for (i = 0; i < length; i++)
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147 out[i] = weight_coeff_a * in_a[i]
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148 + weight_coeff_b * in_b[i];
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149 }
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150
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151 static av_cold int amrnb_decode_init(AVCodecContext *avctx)
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152 {
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153 AMRContext *p = avctx->priv_data;
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154 int i;
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155
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156 avctx->sample_fmt = SAMPLE_FMT_FLT;
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157
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158 // p->excitation always points to the same position in p->excitation_buf
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159 p->excitation = &p->excitation_buf[PITCH_DELAY_MAX + LP_FILTER_ORDER + 1];
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160
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161 for (i = 0; i < LP_FILTER_ORDER; i++) {
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162 p->prev_lsp_sub4[i] = lsp_sub4_init[i] * 1000 / (float)(1 << 15);
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163 p->lsf_avg[i] = p->lsf_q[3][i] = lsp_avg_init[i] / (float)(1 << 15);
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164 }
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165
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166 for (i = 0; i < 4; i++)
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167 p->prediction_error[i] = MIN_ENERGY;
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168
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169 return 0;
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170 }
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171
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172
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173 /**
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174 * Unpack an RFC4867 speech frame into the AMR frame mode and parameters.
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175 *
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176 * The order of speech bits is specified by 3GPP TS 26.101.
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177 *
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178 * @param p the context
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179 * @param buf pointer to the input buffer
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180 * @param buf_size size of the input buffer
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181 *
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182 * @return the frame mode
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183 */
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184 static enum Mode unpack_bitstream(AMRContext *p, const uint8_t *buf,
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185 int buf_size)
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186 {
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187 GetBitContext gb;
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188 enum Mode mode;
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189
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190 init_get_bits(&gb, buf, buf_size * 8);
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191
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192 // Decode the first octet.
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193 skip_bits(&gb, 1); // padding bit
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194 mode = get_bits(&gb, 4); // frame type
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195 p->bad_frame_indicator = !get_bits1(&gb); // quality bit
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196 skip_bits(&gb, 2); // two padding bits
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197
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198 if (mode <= MODE_DTX) {
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199 uint16_t *data = (uint16_t *)&p->frame;
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200 const uint8_t *order = amr_unpacking_bitmaps_per_mode[mode];
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201 int field_size;
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202
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203 memset(&p->frame, 0, sizeof(AMRNBFrame));
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204 buf++;
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205 while ((field_size = *order++)) {
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206 int field = 0;
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207 int field_offset = *order++;
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208 while (field_size--) {
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209 int bit = *order++;
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210 field <<= 1;
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211 field |= buf[bit >> 3] >> (bit & 7) & 1;
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212 }
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213 data[field_offset] = field;
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214 }
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215 }
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216
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217 return mode;
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218 }
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219
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220
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221 /// @defgroup amr_lpc_decoding AMR pitch LPC coefficient decoding functions
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222 /// @{
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223
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224 /**
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225 * Convert an lsf vector into an lsp vector.
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226 *
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227 * @param lsf input lsf vector
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228 * @param lsp output lsp vector
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229 */
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230 static void lsf2lsp(const float *lsf, double *lsp)
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231 {
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232 int i;
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233
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234 for (i = 0; i < LP_FILTER_ORDER; i++)
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235 lsp[i] = cos(2.0 * M_PI * lsf[i]);
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236 }
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237
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238 /**
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239 * Interpolate the LSF vector (used for fixed gain smoothing).
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240 * The interpolation is done over all four subframes even in MODE_12k2.
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241 *
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242 * @param[in,out] lsf_q LSFs in [0,1] for each subframe
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243 * @param[in] lsf_new New LSFs in [0,1] for subframe 4
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244 */
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245 static void interpolate_lsf(float lsf_q[4][LP_FILTER_ORDER], float *lsf_new)
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246 {
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247 int i;
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248
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249 for (i = 0; i < 4; i++)
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250 ff_weighted_vector_sumf(lsf_q[i], lsf_q[3], lsf_new,
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251 0.25 * (3 - i), 0.25 * (i + 1),
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252 LP_FILTER_ORDER);
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253 }
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254
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255 /**
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256 * Decode a set of 5 split-matrix quantized lsf indexes into an lsp vector.
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257 *
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258 * @param p the context
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259 * @param lsp output LSP vector
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260 * @param lsf_no_r LSF vector without the residual vector added
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261 * @param lsf_quantizer pointers to LSF dictionary tables
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262 * @param quantizer_offset offset in tables
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263 * @param sign for the 3 dictionary table
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264 * @param update store data for computing the next frame's LSFs
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265 */
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266 static void lsf2lsp_for_mode12k2(AMRContext *p, double lsp[LP_FILTER_ORDER],
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267 const float lsf_no_r[LP_FILTER_ORDER],
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268 const int16_t *lsf_quantizer[5],
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269 const int quantizer_offset,
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270 const int sign, const int update)
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271 {
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272 int16_t lsf_r[LP_FILTER_ORDER]; // residual LSF vector
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273 float lsf_q[LP_FILTER_ORDER]; // quantified LSF vector
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274 int i;
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275
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276 for (i = 0; i < LP_FILTER_ORDER >> 1; i++)
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277 memcpy(&lsf_r[i << 1], &lsf_quantizer[i][quantizer_offset],
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278 2 * sizeof(*lsf_r));
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279
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280 if (sign) {
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281 lsf_r[4] *= -1;
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282 lsf_r[5] *= -1;
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283 }
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284
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285 if (update)
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286 memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(float));
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287
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288 for (i = 0; i < LP_FILTER_ORDER; i++)
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289 lsf_q[i] = lsf_r[i] * (LSF_R_FAC / 8000.0) + lsf_no_r[i] * (1.0 / 8000.0);
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290
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291 ff_set_min_dist_lsf(lsf_q, MIN_LSF_SPACING, LP_FILTER_ORDER);
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292
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293 if (update)
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294 interpolate_lsf(p->lsf_q, lsf_q);
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295
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296 lsf2lsp(lsf_q, lsp);
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297 }
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298
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299 /**
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300 * Decode a set of 5 split-matrix quantized lsf indexes into 2 lsp vectors.
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301 *
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302 * @param p pointer to the AMRContext
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303 */
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304 static void lsf2lsp_5(AMRContext *p)
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305 {
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306 const uint16_t *lsf_param = p->frame.lsf;
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307 float lsf_no_r[LP_FILTER_ORDER]; // LSFs without the residual vector
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308 const int16_t *lsf_quantizer[5];
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309 int i;
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310
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311 lsf_quantizer[0] = lsf_5_1[lsf_param[0]];
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312 lsf_quantizer[1] = lsf_5_2[lsf_param[1]];
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313 lsf_quantizer[2] = lsf_5_3[lsf_param[2] >> 1];
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314 lsf_quantizer[3] = lsf_5_4[lsf_param[3]];
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315 lsf_quantizer[4] = lsf_5_5[lsf_param[4]];
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316
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317 for (i = 0; i < LP_FILTER_ORDER; i++)
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318 lsf_no_r[i] = p->prev_lsf_r[i] * LSF_R_FAC * PRED_FAC_MODE_12k2 + lsf_5_mean[i];
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319
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320 lsf2lsp_for_mode12k2(p, p->lsp[1], lsf_no_r, lsf_quantizer, 0, lsf_param[2] & 1, 0);
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321 lsf2lsp_for_mode12k2(p, p->lsp[3], lsf_no_r, lsf_quantizer, 2, lsf_param[2] & 1, 1);
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322
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323 // interpolate LSP vectors at subframes 1 and 3
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324 weighted_vector_sumd(p->lsp[0], p->prev_lsp_sub4, p->lsp[1], 0.5, 0.5, LP_FILTER_ORDER);
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325 weighted_vector_sumd(p->lsp[2], p->lsp[1] , p->lsp[3], 0.5, 0.5, LP_FILTER_ORDER);
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326 }
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327
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328 /**
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329 * Decode a set of 3 split-matrix quantized lsf indexes into an lsp vector.
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330 *
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331 * @param p pointer to the AMRContext
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332 */
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333 static void lsf2lsp_3(AMRContext *p)
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334 {
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335 const uint16_t *lsf_param = p->frame.lsf;
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336 int16_t lsf_r[LP_FILTER_ORDER]; // residual LSF vector
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337 float lsf_q[LP_FILTER_ORDER]; // quantified LSF vector
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338 const int16_t *lsf_quantizer;
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339 int i, j;
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340
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341 lsf_quantizer = (p->cur_frame_mode == MODE_7k95 ? lsf_3_1_MODE_7k95 : lsf_3_1)[lsf_param[0]];
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342 memcpy(lsf_r, lsf_quantizer, 3 * sizeof(*lsf_r));
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343
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344 lsf_quantizer = lsf_3_2[lsf_param[1] << (p->cur_frame_mode <= MODE_5k15)];
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345 memcpy(lsf_r + 3, lsf_quantizer, 3 * sizeof(*lsf_r));
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346
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347 lsf_quantizer = (p->cur_frame_mode <= MODE_5k15 ? lsf_3_3_MODE_5k15 : lsf_3_3)[lsf_param[2]];
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348 memcpy(lsf_r + 6, lsf_quantizer, 4 * sizeof(*lsf_r));
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349
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350 // calculate mean-removed LSF vector and add mean
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351 for (i = 0; i < LP_FILTER_ORDER; i++)
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352 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);
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353
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354 ff_set_min_dist_lsf(lsf_q, MIN_LSF_SPACING, LP_FILTER_ORDER);
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355
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356 // store data for computing the next frame's LSFs
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357 interpolate_lsf(p->lsf_q, lsf_q);
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358 memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(*lsf_r));
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359
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360 lsf2lsp(lsf_q, p->lsp[3]);
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361
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362 // interpolate LSP vectors at subframes 1, 2 and 3
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363 for (i = 1; i <= 3; i++)
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364 for(j = 0; j < LP_FILTER_ORDER; j++)
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365 p->lsp[i-1][j] = p->prev_lsp_sub4[j] +
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366 (p->lsp[3][j] - p->prev_lsp_sub4[j]) * 0.25 * i;
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367 }
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368
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369 /// @}
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370
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371
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372 /// @defgroup amr_pitch_vector_decoding AMR pitch vector decoding functions
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373 /// @{
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374
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375 /**
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376 * Like ff_decode_pitch_lag(), but with 1/6 resolution
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377 */
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378 static void decode_pitch_lag_1_6(int *lag_int, int *lag_frac, int pitch_index,
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379 const int prev_lag_int, const int subframe)
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380 {
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381 if (subframe == 0 || subframe == 2) {
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382 if (pitch_index < 463) {
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383 *lag_int = (pitch_index + 107) * 10923 >> 16;
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384 *lag_frac = pitch_index - *lag_int * 6 + 105;
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385 } else {
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386 *lag_int = pitch_index - 368;
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387 *lag_frac = 0;
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388 }
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389 } else {
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390 *lag_int = ((pitch_index + 5) * 10923 >> 16) - 1;
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391 *lag_frac = pitch_index - *lag_int * 6 - 3;
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392 *lag_int += av_clip(prev_lag_int - 5, PITCH_LAG_MIN_MODE_12k2,
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393 PITCH_DELAY_MAX - 9);
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394 }
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395 }
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396
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397 static void decode_pitch_vector(AMRContext *p,
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398 const AMRNBSubframe *amr_subframe,
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399 const int subframe)
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400 {
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401 int pitch_lag_int, pitch_lag_frac;
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402 enum Mode mode = p->cur_frame_mode;
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403
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404 if (p->cur_frame_mode == MODE_12k2) {
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405 decode_pitch_lag_1_6(&pitch_lag_int, &pitch_lag_frac,
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406 amr_subframe->p_lag, p->pitch_lag_int,
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407 subframe);
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408 } else
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409 ff_decode_pitch_lag(&pitch_lag_int, &pitch_lag_frac,
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410 amr_subframe->p_lag,
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411 p->pitch_lag_int, subframe,
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412 mode != MODE_4k75 && mode != MODE_5k15,
|
|
413 mode <= MODE_6k7 ? 4 : (mode == MODE_7k95 ? 5 : 6));
|
|
414
|
|
415 p->pitch_lag_int = pitch_lag_int; // store previous lag in a uint8_t
|
|
416
|
|
417 pitch_lag_frac <<= (p->cur_frame_mode != MODE_12k2);
|
|
418
|
|
419 pitch_lag_int += pitch_lag_frac > 0;
|
|
420
|
|
421 /* Calculate the pitch vector by interpolating the past excitation at the
|
|
422 pitch lag using a b60 hamming windowed sinc function. */
|
|
423 ff_acelp_interpolatef(p->excitation, p->excitation + 1 - pitch_lag_int,
|
|
424 ff_b60_sinc, 6,
|
|
425 pitch_lag_frac + 6 - 6*(pitch_lag_frac > 0),
|
|
426 10, AMR_SUBFRAME_SIZE);
|
|
427
|
|
428 memcpy(p->pitch_vector, p->excitation, AMR_SUBFRAME_SIZE * sizeof(float));
|
|
429 }
|
|
430
|
|
431 /// @}
|
|
432
|
|
433
|
|
434 /// @defgroup amr_algebraic_code_book AMR algebraic code book (fixed) vector decoding functions
|
|
435 /// @{
|
|
436
|
|
437 /**
|
|
438 * Decode a 10-bit algebraic codebook index from a 10.2 kbit/s frame.
|
|
439 */
|
|
440 static void decode_10bit_pulse(int code, int pulse_position[8],
|
|
441 int i1, int i2, int i3)
|
|
442 {
|
|
443 // coded using 7+3 bits with the 3 LSBs being, individually, the LSB of 1 of
|
|
444 // the 3 pulses and the upper 7 bits being coded in base 5
|
|
445 const uint8_t *positions = base_five_table[code >> 3];
|
|
446 pulse_position[i1] = (positions[2] << 1) + ( code & 1);
|
|
447 pulse_position[i2] = (positions[1] << 1) + ((code >> 1) & 1);
|
|
448 pulse_position[i3] = (positions[0] << 1) + ((code >> 2) & 1);
|
|
449 }
|
|
450
|
|
451 /**
|
|
452 * Decode the algebraic codebook index to pulse positions and signs and
|
|
453 * construct the algebraic codebook vector for MODE_10k2.
|
|
454 *
|
|
455 * @param fixed_index positions of the eight pulses
|
|
456 * @param fixed_sparse pointer to the algebraic codebook vector
|
|
457 */
|
|
458 static void decode_8_pulses_31bits(const int16_t *fixed_index,
|
|
459 AMRFixed *fixed_sparse)
|
|
460 {
|
|
461 int pulse_position[8];
|
|
462 int i, temp;
|
|
463
|
|
464 decode_10bit_pulse(fixed_index[4], pulse_position, 0, 4, 1);
|
|
465 decode_10bit_pulse(fixed_index[5], pulse_position, 2, 6, 5);
|
|
466
|
|
467 // coded using 5+2 bits with the 2 LSBs being, individually, the LSB of 1 of
|
|
468 // the 2 pulses and the upper 5 bits being coded in base 5
|
|
469 temp = ((fixed_index[6] >> 2) * 25 + 12) >> 5;
|
|
470 pulse_position[3] = temp % 5;
|
|
471 pulse_position[7] = temp / 5;
|
|
472 if (pulse_position[7] & 1)
|
|
473 pulse_position[3] = 4 - pulse_position[3];
|
|
474 pulse_position[3] = (pulse_position[3] << 1) + ( fixed_index[6] & 1);
|
|
475 pulse_position[7] = (pulse_position[7] << 1) + ((fixed_index[6] >> 1) & 1);
|
|
476
|
|
477 fixed_sparse->n = 8;
|
|
478 for (i = 0; i < 4; i++) {
|
|
479 const int pos1 = (pulse_position[i] << 2) + i;
|
|
480 const int pos2 = (pulse_position[i + 4] << 2) + i;
|
|
481 const float sign = fixed_index[i] ? -1.0 : 1.0;
|
|
482 fixed_sparse->x[i ] = pos1;
|
|
483 fixed_sparse->x[i + 4] = pos2;
|
|
484 fixed_sparse->y[i ] = sign;
|
|
485 fixed_sparse->y[i + 4] = pos2 < pos1 ? -sign : sign;
|
|
486 }
|
|
487 }
|
|
488
|
|
489 /**
|
|
490 * Decode the algebraic codebook index to pulse positions and signs,
|
|
491 * then construct the algebraic codebook vector.
|
|
492 *
|
|
493 * nb of pulses | bits encoding pulses
|
|
494 * For MODE_4k75 or MODE_5k15, 2 | 1-3, 4-6, 7
|
|
495 * MODE_5k9, 2 | 1, 2-4, 5-6, 7-9
|
|
496 * MODE_6k7, 3 | 1-3, 4, 5-7, 8, 9-11
|
|
497 * MODE_7k4 or MODE_7k95, 4 | 1-3, 4-6, 7-9, 10, 11-13
|
|
498 *
|
|
499 * @param fixed_sparse pointer to the algebraic codebook vector
|
|
500 * @param pulses algebraic codebook indexes
|
|
501 * @param mode mode of the current frame
|
|
502 * @param subframe current subframe number
|
|
503 */
|
|
504 static void decode_fixed_sparse(AMRFixed *fixed_sparse, const uint16_t *pulses,
|
|
505 const enum Mode mode, const int subframe)
|
|
506 {
|
|
507 assert(MODE_4k75 <= mode && mode <= MODE_12k2);
|
|
508
|
|
509 if (mode == MODE_12k2) {
|
|
510 ff_decode_10_pulses_35bits(pulses, fixed_sparse, gray_decode, 5, 3);
|
|
511 } else if (mode == MODE_10k2) {
|
|
512 decode_8_pulses_31bits(pulses, fixed_sparse);
|
|
513 } else {
|
|
514 int *pulse_position = fixed_sparse->x;
|
|
515 int i, pulse_subset;
|
|
516 const int fixed_index = pulses[0];
|
|
517
|
|
518 if (mode <= MODE_5k15) {
|
|
519 pulse_subset = ((fixed_index >> 3) & 8) + (subframe << 1);
|
|
520 pulse_position[0] = ( fixed_index & 7) * 5 + track_position[pulse_subset];
|
|
521 pulse_position[1] = ((fixed_index >> 3) & 7) * 5 + track_position[pulse_subset + 1];
|
|
522 fixed_sparse->n = 2;
|
|
523 } else if (mode == MODE_5k9) {
|
|
524 pulse_subset = ((fixed_index & 1) << 1) + 1;
|
|
525 pulse_position[0] = ((fixed_index >> 1) & 7) * 5 + pulse_subset;
|
|
526 pulse_subset = (fixed_index >> 4) & 3;
|
|
527 pulse_position[1] = ((fixed_index >> 6) & 7) * 5 + pulse_subset + (pulse_subset == 3 ? 1 : 0);
|
|
528 fixed_sparse->n = pulse_position[0] == pulse_position[1] ? 1 : 2;
|
|
529 } else if (mode == MODE_6k7) {
|
|
530 pulse_position[0] = (fixed_index & 7) * 5;
|
|
531 pulse_subset = (fixed_index >> 2) & 2;
|
|
532 pulse_position[1] = ((fixed_index >> 4) & 7) * 5 + pulse_subset + 1;
|
|
533 pulse_subset = (fixed_index >> 6) & 2;
|
|
534 pulse_position[2] = ((fixed_index >> 8) & 7) * 5 + pulse_subset + 2;
|
|
535 fixed_sparse->n = 3;
|
|
536 } else { // mode <= MODE_7k95
|
|
537 pulse_position[0] = gray_decode[ fixed_index & 7];
|
|
538 pulse_position[1] = gray_decode[(fixed_index >> 3) & 7] + 1;
|
|
539 pulse_position[2] = gray_decode[(fixed_index >> 6) & 7] + 2;
|
|
540 pulse_subset = (fixed_index >> 9) & 1;
|
|
541 pulse_position[3] = gray_decode[(fixed_index >> 10) & 7] + pulse_subset + 3;
|
|
542 fixed_sparse->n = 4;
|
|
543 }
|
|
544 for (i = 0; i < fixed_sparse->n; i++)
|
|
545 fixed_sparse->y[i] = (pulses[1] >> i) & 1 ? 1.0 : -1.0;
|
|
546 }
|
|
547 }
|
|
548
|
|
549 /**
|
|
550 * Apply pitch lag to obtain the sharpened fixed vector (section 6.1.2)
|
|
551 *
|
|
552 * @param p the context
|
|
553 * @param subframe unpacked amr subframe
|
|
554 * @param mode mode of the current frame
|
|
555 * @param fixed_sparse sparse respresentation of the fixed vector
|
|
556 */
|
|
557 static void pitch_sharpening(AMRContext *p, int subframe, enum Mode mode,
|
|
558 AMRFixed *fixed_sparse)
|
|
559 {
|
|
560 // The spec suggests the current pitch gain is always used, but in other
|
|
561 // modes the pitch and codebook gains are joinly quantized (sec 5.8.2)
|
|
562 // so the codebook gain cannot depend on the quantized pitch gain.
|
|
563 if (mode == MODE_12k2)
|
|
564 p->beta = FFMIN(p->pitch_gain[4], 1.0);
|
|
565
|
|
566 fixed_sparse->pitch_lag = p->pitch_lag_int;
|
|
567 fixed_sparse->pitch_fac = p->beta;
|
|
568
|
|
569 // Save pitch sharpening factor for the next subframe
|
|
570 // MODE_4k75 only updates on the 2nd and 4th subframes - this follows from
|
|
571 // the fact that the gains for two subframes are jointly quantized.
|
|
572 if (mode != MODE_4k75 || subframe & 1)
|
|
573 p->beta = av_clipf(p->pitch_gain[4], 0.0, SHARP_MAX);
|
|
574 }
|
|
575 /// @}
|
|
576
|
|
577
|
|
578 /// @defgroup amr_gain_decoding AMR gain decoding functions
|
|
579 /// @{
|
|
580
|
|
581 /**
|
|
582 * fixed gain smoothing
|
|
583 * Note that where the spec specifies the "spectrum in the q domain"
|
|
584 * in section 6.1.4, in fact frequencies should be used.
|
|
585 *
|
|
586 * @param p the context
|
|
587 * @param lsf LSFs for the current subframe, in the range [0,1]
|
|
588 * @param lsf_avg averaged LSFs
|
|
589 * @param mode mode of the current frame
|
|
590 *
|
|
591 * @return fixed gain smoothed
|
|
592 */
|
|
593 static float fixed_gain_smooth(AMRContext *p , const float *lsf,
|
|
594 const float *lsf_avg, const enum Mode mode)
|
|
595 {
|
|
596 float diff = 0.0;
|
|
597 int i;
|
|
598
|
|
599 for (i = 0; i < LP_FILTER_ORDER; i++)
|
|
600 diff += fabs(lsf_avg[i] - lsf[i]) / lsf_avg[i];
|
|
601
|
|
602 // If diff is large for ten subframes, disable smoothing for a 40-subframe
|
|
603 // hangover period.
|
|
604 p->diff_count++;
|
|
605 if (diff <= 0.65)
|
|
606 p->diff_count = 0;
|
|
607
|
|
608 if (p->diff_count > 10) {
|
|
609 p->hang_count = 0;
|
|
610 p->diff_count--; // don't let diff_count overflow
|
|
611 }
|
|
612
|
|
613 if (p->hang_count < 40) {
|
|
614 p->hang_count++;
|
|
615 } else if (mode < MODE_7k4 || mode == MODE_10k2) {
|
|
616 const float smoothing_factor = av_clipf(4.0 * diff - 1.6, 0.0, 1.0);
|
|
617 const float fixed_gain_mean = (p->fixed_gain[0] + p->fixed_gain[1] +
|
|
618 p->fixed_gain[2] + p->fixed_gain[3] +
|
|
619 p->fixed_gain[4]) * 0.2;
|
|
620 return smoothing_factor * p->fixed_gain[4] +
|
|
621 (1.0 - smoothing_factor) * fixed_gain_mean;
|
|
622 }
|
|
623 return p->fixed_gain[4];
|
|
624 }
|
|
625
|
|
626 /**
|
|
627 * Decode pitch gain and fixed gain factor (part of section 6.1.3).
|
|
628 *
|
|
629 * @param p the context
|
|
630 * @param amr_subframe unpacked amr subframe
|
|
631 * @param mode mode of the current frame
|
|
632 * @param subframe current subframe number
|
|
633 * @param fixed_gain_factor decoded gain correction factor
|
|
634 */
|
|
635 static void decode_gains(AMRContext *p, const AMRNBSubframe *amr_subframe,
|
|
636 const enum Mode mode, const int subframe,
|
|
637 float *fixed_gain_factor)
|
|
638 {
|
|
639 if (mode == MODE_12k2 || mode == MODE_7k95) {
|
|
640 p->pitch_gain[4] = qua_gain_pit [amr_subframe->p_gain ]
|
|
641 * (1.0 / 16384.0);
|
|
642 *fixed_gain_factor = qua_gain_code[amr_subframe->fixed_gain]
|
|
643 * (1.0 / 2048.0);
|
|
644 } else {
|
|
645 const uint16_t *gains;
|
|
646
|
|
647 if (mode >= MODE_6k7) {
|
|
648 gains = gains_high[amr_subframe->p_gain];
|
|
649 } else if (mode >= MODE_5k15) {
|
|
650 gains = gains_low [amr_subframe->p_gain];
|
|
651 } else {
|
|
652 // gain index is only coded in subframes 0,2 for MODE_4k75
|
|
653 gains = gains_MODE_4k75[(p->frame.subframe[subframe & 2].p_gain << 1) + (subframe & 1)];
|
|
654 }
|
|
655
|
|
656 p->pitch_gain[4] = gains[0] * (1.0 / 16384.0);
|
|
657 *fixed_gain_factor = gains[1] * (1.0 / 4096.0);
|
|
658 }
|
|
659 }
|
|
660
|
|
661 /// @}
|
|
662
|
|
663
|
|
664 /// @defgroup amr_pre_processing AMR pre-processing functions
|
|
665 /// @{
|
|
666
|
|
667 /**
|
|
668 * Circularly convolve a sparse fixed vector with a phase dispersion impulse
|
|
669 * response filter (D.6.2 of G.729 and 6.1.5 of AMR).
|
|
670 *
|
|
671 * @param out vector with filter applied
|
|
672 * @param in source vector
|
|
673 * @param filter phase filter coefficients
|
|
674 *
|
|
675 * out[n] = sum(i,0,len-1){ in[i] * filter[(len + n - i)%len] }
|
|
676 */
|
|
677 static void apply_ir_filter(float *out, const AMRFixed *in,
|
|
678 const float *filter)
|
|
679 {
|
|
680 float filter1[AMR_SUBFRAME_SIZE], //!< filters at pitch lag*1 and *2
|
|
681 filter2[AMR_SUBFRAME_SIZE];
|
|
682 int lag = in->pitch_lag;
|
|
683 float fac = in->pitch_fac;
|
|
684 int i;
|
|
685
|
|
686 if (lag < AMR_SUBFRAME_SIZE) {
|
|
687 ff_celp_circ_addf(filter1, filter, filter, lag, fac,
|
|
688 AMR_SUBFRAME_SIZE);
|
|
689
|
|
690 if (lag < AMR_SUBFRAME_SIZE >> 1)
|
|
691 ff_celp_circ_addf(filter2, filter, filter1, lag, fac,
|
|
692 AMR_SUBFRAME_SIZE);
|
|
693 }
|
|
694
|
|
695 memset(out, 0, sizeof(float) * AMR_SUBFRAME_SIZE);
|
|
696 for (i = 0; i < in->n; i++) {
|
|
697 int x = in->x[i];
|
|
698 float y = in->y[i];
|
|
699 const float *filterp;
|
|
700
|
|
701 if (x >= AMR_SUBFRAME_SIZE - lag) {
|
|
702 filterp = filter;
|
|
703 } else if (x >= AMR_SUBFRAME_SIZE - (lag << 1)) {
|
|
704 filterp = filter1;
|
|
705 } else
|
|
706 filterp = filter2;
|
|
707
|
|
708 ff_celp_circ_addf(out, out, filterp, x, y, AMR_SUBFRAME_SIZE);
|
|
709 }
|
|
710 }
|
|
711
|
|
712 /**
|
|
713 * Reduce fixed vector sparseness by smoothing with one of three IR filters.
|
|
714 * Also know as "adaptive phase dispersion".
|
|
715 *
|
|
716 * This implements 3GPP TS 26.090 section 6.1(5).
|
|
717 *
|
|
718 * @param p the context
|
|
719 * @param fixed_sparse algebraic codebook vector
|
|
720 * @param fixed_vector unfiltered fixed vector
|
|
721 * @param fixed_gain smoothed gain
|
|
722 * @param out space for modified vector if necessary
|
|
723 */
|
|
724 static const float *anti_sparseness(AMRContext *p, AMRFixed *fixed_sparse,
|
|
725 const float *fixed_vector,
|
|
726 float fixed_gain, float *out)
|
|
727 {
|
|
728 int ir_filter_nr;
|
|
729
|
|
730 if (p->pitch_gain[4] < 0.6) {
|
|
731 ir_filter_nr = 0; // strong filtering
|
|
732 } else if (p->pitch_gain[4] < 0.9) {
|
|
733 ir_filter_nr = 1; // medium filtering
|
|
734 } else
|
|
735 ir_filter_nr = 2; // no filtering
|
|
736
|
|
737 // detect 'onset'
|
|
738 if (fixed_gain > 2.0 * p->prev_sparse_fixed_gain) {
|
|
739 p->ir_filter_onset = 2;
|
|
740 } else if (p->ir_filter_onset)
|
|
741 p->ir_filter_onset--;
|
|
742
|
|
743 if (!p->ir_filter_onset) {
|
|
744 int i, count = 0;
|
|
745
|
|
746 for (i = 0; i < 5; i++)
|
|
747 if (p->pitch_gain[i] < 0.6)
|
|
748 count++;
|
|
749 if (count > 2)
|
|
750 ir_filter_nr = 0;
|
|
751
|
|
752 if (ir_filter_nr > p->prev_ir_filter_nr + 1)
|
|
753 ir_filter_nr--;
|
|
754 } else if (ir_filter_nr < 2)
|
|
755 ir_filter_nr++;
|
|
756
|
|
757 // Disable filtering for very low level of fixed_gain.
|
|
758 // Note this step is not specified in the technical description but is in
|
|
759 // the reference source in the function Ph_disp.
|
|
760 if (fixed_gain < 5.0)
|
|
761 ir_filter_nr = 2;
|
|
762
|
|
763 if (p->cur_frame_mode != MODE_7k4 && p->cur_frame_mode < MODE_10k2
|
|
764 && ir_filter_nr < 2) {
|
|
765 apply_ir_filter(out, fixed_sparse,
|
|
766 (p->cur_frame_mode == MODE_7k95 ?
|
|
767 ir_filters_lookup_MODE_7k95 :
|
|
768 ir_filters_lookup)[ir_filter_nr]);
|
|
769 fixed_vector = out;
|
|
770 }
|
|
771
|
|
772 // update ir filter strength history
|
|
773 p->prev_ir_filter_nr = ir_filter_nr;
|
|
774 p->prev_sparse_fixed_gain = fixed_gain;
|
|
775
|
|
776 return fixed_vector;
|
|
777 }
|
|
778
|
|
779 /// @}
|
|
780
|
|
781
|
|
782 /// @defgroup amr_synthesis AMR synthesis functions
|
|
783 /// @{
|
|
784
|
|
785 /**
|
|
786 * Conduct 10th order linear predictive coding synthesis.
|
|
787 *
|
|
788 * @param p pointer to the AMRContext
|
|
789 * @param lpc pointer to the LPC coefficients
|
|
790 * @param fixed_gain fixed codebook gain for synthesis
|
|
791 * @param fixed_vector algebraic codebook vector
|
|
792 * @param samples pointer to the output speech samples
|
|
793 * @param overflow 16-bit overflow flag
|
|
794 */
|
|
795 static int synthesis(AMRContext *p, float *lpc,
|
|
796 float fixed_gain, const float *fixed_vector,
|
|
797 float *samples, uint8_t overflow)
|
|
798 {
|
|
799 int i, overflow_temp = 0;
|
|
800 float excitation[AMR_SUBFRAME_SIZE];
|
|
801
|
|
802 // if an overflow has been detected, the pitch vector is scaled down by a
|
|
803 // factor of 4
|
|
804 if (overflow)
|
|
805 for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
|
|
806 p->pitch_vector[i] *= 0.25;
|
|
807
|
|
808 ff_weighted_vector_sumf(excitation, p->pitch_vector, fixed_vector,
|
|
809 p->pitch_gain[4], fixed_gain, AMR_SUBFRAME_SIZE);
|
|
810
|
|
811 // emphasize pitch vector contribution
|
|
812 if (p->pitch_gain[4] > 0.5 && !overflow) {
|
|
813 float energy = ff_dot_productf(excitation, excitation,
|
|
814 AMR_SUBFRAME_SIZE);
|
|
815 float pitch_factor =
|
|
816 p->pitch_gain[4] *
|
|
817 (p->cur_frame_mode == MODE_12k2 ?
|
|
818 0.25 * FFMIN(p->pitch_gain[4], 1.0) :
|
|
819 0.5 * FFMIN(p->pitch_gain[4], SHARP_MAX));
|
|
820
|
|
821 for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
|
|
822 excitation[i] += pitch_factor * p->pitch_vector[i];
|
|
823
|
|
824 ff_scale_vector_to_given_sum_of_squares(excitation, excitation, energy,
|
|
825 AMR_SUBFRAME_SIZE);
|
|
826 }
|
|
827
|
|
828 ff_celp_lp_synthesis_filterf(samples, lpc, excitation, AMR_SUBFRAME_SIZE,
|
|
829 LP_FILTER_ORDER);
|
|
830
|
|
831 // detect overflow
|
|
832 for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
|
|
833 if (fabsf(samples[i]) > AMR_SAMPLE_BOUND) {
|
|
834 overflow_temp = 1;
|
|
835 samples[i] = av_clipf(samples[i], -AMR_SAMPLE_BOUND,
|
|
836 AMR_SAMPLE_BOUND);
|
|
837 }
|
|
838
|
|
839 return overflow_temp;
|
|
840 }
|
|
841
|
|
842 /// @}
|
|
843
|
|
844
|
|
845 /// @defgroup amr_update AMR update functions
|
|
846 /// @{
|
|
847
|
|
848 /**
|
|
849 * Update buffers and history at the end of decoding a subframe.
|
|
850 *
|
|
851 * @param p pointer to the AMRContext
|
|
852 */
|
|
853 static void update_state(AMRContext *p)
|
|
854 {
|
|
855 memcpy(p->prev_lsp_sub4, p->lsp[3], LP_FILTER_ORDER * sizeof(p->lsp[3][0]));
|
|
856
|
|
857 memmove(&p->excitation_buf[0], &p->excitation_buf[AMR_SUBFRAME_SIZE],
|
|
858 (PITCH_DELAY_MAX + LP_FILTER_ORDER + 1) * sizeof(float));
|
|
859
|
|
860 memmove(&p->pitch_gain[0], &p->pitch_gain[1], 4 * sizeof(float));
|
|
861 memmove(&p->fixed_gain[0], &p->fixed_gain[1], 4 * sizeof(float));
|
|
862
|
|
863 memmove(&p->samples_in[0], &p->samples_in[AMR_SUBFRAME_SIZE],
|
|
864 LP_FILTER_ORDER * sizeof(float));
|
|
865 }
|
|
866
|
|
867 /// @}
|
|
868
|
|
869
|
|
870 /// @defgroup amr_postproc AMR Post processing functions
|
|
871 /// @{
|
|
872
|
|
873 /**
|
|
874 * Get the tilt factor of a formant filter from its transfer function
|
|
875 *
|
|
876 * @param lpc_n LP_FILTER_ORDER coefficients of the numerator
|
|
877 * @param lpc_d LP_FILTER_ORDER coefficients of the denominator
|
|
878 */
|
|
879 static float tilt_factor(float *lpc_n, float *lpc_d)
|
|
880 {
|
|
881 float rh0, rh1; // autocorrelation at lag 0 and 1
|
|
882
|
|
883 // LP_FILTER_ORDER prior zeros are needed for ff_celp_lp_synthesis_filterf
|
|
884 float impulse_buffer[LP_FILTER_ORDER + AMR_TILT_RESPONSE] = { 0 };
|
|
885 float *hf = impulse_buffer + LP_FILTER_ORDER; // start of impulse response
|
|
886
|
|
887 hf[0] = 1.0;
|
|
888 memcpy(hf + 1, lpc_n, sizeof(float) * LP_FILTER_ORDER);
|
|
889 ff_celp_lp_synthesis_filterf(hf, lpc_d, hf, AMR_TILT_RESPONSE,
|
|
890 LP_FILTER_ORDER);
|
|
891
|
|
892 rh0 = ff_dot_productf(hf, hf, AMR_TILT_RESPONSE);
|
|
893 rh1 = ff_dot_productf(hf, hf + 1, AMR_TILT_RESPONSE - 1);
|
|
894
|
|
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;
|
|
898 }
|
|
899
|
|
900 /**
|
|
901 * Perform adaptive post-filtering to enhance the quality of the speech.
|
|
902 * See section 6.2.1.
|
|
903 *
|
|
904 * @param p pointer to the AMRContext
|
|
905 * @param lpc interpolated LP coefficients for this subframe
|
|
906 * @param buf_out output of the filter
|
|
907 */
|
|
908 static void postfilter(AMRContext *p, float *lpc, float *buf_out)
|
|
909 {
|
|
910 int i;
|
|
911 float *samples = p->samples_in + LP_FILTER_ORDER; // Start of input
|
|
912
|
|
913 float speech_gain = ff_dot_productf(samples, samples,
|
|
914 AMR_SUBFRAME_SIZE);
|
|
915
|
|
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
|
|
919
|
|
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;
|
|
923 } else {
|
|
924 gamma_n = ff_pow_0_55;
|
|
925 gamma_d = ff_pow_0_7;
|
|
926 }
|
|
927
|
|
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];
|
|
931 }
|
|
932
|
|
933 memcpy(pole_out, p->postfilter_mem, sizeof(float) * LP_FILTER_ORDER);
|
|
934 ff_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);
|
|
938
|
|
939 ff_celp_lp_zero_synthesis_filterf(buf_out, lpc_n,
|
|
940 pole_out + LP_FILTER_ORDER,
|
|
941 AMR_SUBFRAME_SIZE, LP_FILTER_ORDER);
|
|
942
|
|
943 ff_tilt_compensation(&p->tilt_mem, tilt_factor(lpc_n, lpc_d), buf_out,
|
|
944 AMR_SUBFRAME_SIZE);
|
|
945
|
|
946 ff_adaptative_gain_control(buf_out, speech_gain, AMR_SUBFRAME_SIZE,
|
|
947 AMR_AGC_ALPHA, &p->postfilter_agc);
|
|
948 }
|
|
949
|
|
950 /// @}
|
|
951
|
|
952 static int amrnb_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
|
|
953 AVPacket *avpkt)
|
|
954 {
|
|
955
|
|
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 = data; // pointer to the output data buffer
|
|
960 int i, subframe;
|
|
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
|
|
966
|
|
967 p->cur_frame_mode = unpack_bitstream(p, buf, buf_size);
|
|
968 if (p->cur_frame_mode == MODE_DTX) {
|
|
969 av_log_missing_feature(avctx, "dtx mode", 1);
|
|
970 return -1;
|
|
971 }
|
|
972
|
|
973 if (p->cur_frame_mode == MODE_12k2) {
|
|
974 lsf2lsp_5(p);
|
|
975 } else
|
|
976 lsf2lsp_3(p);
|
|
977
|
|
978 for (i = 0; i < 4; i++)
|
|
979 ff_acelp_lspd2lpc(p->lsp[i], p->lpc[i], 5);
|
|
980
|
|
981 for (subframe = 0; subframe < 4; subframe++) {
|
|
982 const AMRNBSubframe *amr_subframe = &p->frame.subframe[subframe];
|
|
983
|
|
984 decode_pitch_vector(p, amr_subframe, subframe);
|
|
985
|
|
986 decode_fixed_sparse(&fixed_sparse, amr_subframe->pulses,
|
|
987 p->cur_frame_mode, subframe);
|
|
988
|
|
989 // The fixed gain (section 6.1.3) depends on the fixed vector
|
|
990 // (section 6.1.2), but the fixed vector calculation uses
|
|
991 // pitch sharpening based on the on the pitch gain (section 6.1.3).
|
|
992 // So the correct order is: pitch gain, pitch sharpening, fixed gain.
|
|
993 decode_gains(p, amr_subframe, p->cur_frame_mode, subframe,
|
|
994 &fixed_gain_factor);
|
|
995
|
|
996 pitch_sharpening(p, subframe, p->cur_frame_mode, &fixed_sparse);
|
|
997
|
|
998 ff_set_fixed_vector(p->fixed_vector, &fixed_sparse, 1.0,
|
|
999 AMR_SUBFRAME_SIZE);
|
|
1000
|
|
1001 p->fixed_gain[4] =
|
|
1002 ff_amr_set_fixed_gain(fixed_gain_factor,
|
|
1003 ff_dot_productf(p->fixed_vector, p->fixed_vector,
|
|
1004 AMR_SUBFRAME_SIZE)/AMR_SUBFRAME_SIZE,
|
|
1005 p->prediction_error,
|
|
1006 energy_mean[p->cur_frame_mode], energy_pred_fac);
|
|
1007
|
|
1008 // The excitation feedback is calculated without any processing such
|
|
1009 // as fixed gain smoothing. This isn't mentioned in the specification.
|
|
1010 for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
|
|
1011 p->excitation[i] *= p->pitch_gain[4];
|
|
1012 ff_set_fixed_vector(p->excitation, &fixed_sparse, p->fixed_gain[4],
|
|
1013 AMR_SUBFRAME_SIZE);
|
|
1014
|
|
1015 // In the ref decoder, excitation is stored with no fractional bits.
|
|
1016 // This step prevents buzz in silent periods. The ref encoder can
|
|
1017 // emit long sequences with pitch factor greater than one. This
|
|
1018 // creates unwanted feedback if the excitation vector is nonzero.
|
|
1019 // (e.g. test sequence T19_795.COD in 3GPP TS 26.074)
|
|
1020 for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
|
|
1021 p->excitation[i] = truncf(p->excitation[i]);
|
|
1022
|
|
1023 // Smooth fixed gain.
|
|
1024 // The specification is ambiguous, but in the reference source, the
|
|
1025 // smoothed value is NOT fed back into later fixed gain smoothing.
|
|
1026 synth_fixed_gain = fixed_gain_smooth(p, p->lsf_q[subframe],
|
|
1027 p->lsf_avg, p->cur_frame_mode);
|
|
1028
|
|
1029 synth_fixed_vector = anti_sparseness(p, &fixed_sparse, p->fixed_vector,
|
|
1030 synth_fixed_gain, spare_vector);
|
|
1031
|
|
1032 if (synthesis(p, p->lpc[subframe], synth_fixed_gain,
|
|
1033 synth_fixed_vector, &p->samples_in[LP_FILTER_ORDER], 0))
|
|
1034 // overflow detected -> rerun synthesis scaling pitch vector down
|
|
1035 // by a factor of 4, skipping pitch vector contribution emphasis
|
|
1036 // and adaptive gain control
|
|
1037 synthesis(p, p->lpc[subframe], synth_fixed_gain,
|
|
1038 synth_fixed_vector, &p->samples_in[LP_FILTER_ORDER], 1);
|
|
1039
|
|
1040 postfilter(p, p->lpc[subframe], buf_out + subframe * AMR_SUBFRAME_SIZE);
|
|
1041
|
|
1042 // update buffers and history
|
|
1043 ff_clear_fixed_vector(p->fixed_vector, &fixed_sparse, AMR_SUBFRAME_SIZE);
|
|
1044 update_state(p);
|
|
1045 }
|
|
1046
|
|
1047 ff_acelp_apply_order_2_transfer_function(buf_out, highpass_zeros,
|
|
1048 highpass_poles, highpass_gain,
|
|
1049 p->high_pass_mem, AMR_BLOCK_SIZE);
|
|
1050
|
|
1051 for (i = 0; i < AMR_BLOCK_SIZE; i++)
|
|
1052 buf_out[i] = av_clipf(buf_out[i] * AMR_SAMPLE_SCALE,
|
|
1053 -1.0, 32767.0 / 32768.0);
|
|
1054
|
|
1055 /* Update averaged lsf vector (used for fixed gain smoothing).
|
|
1056 *
|
|
1057 * Note that lsf_avg should not incorporate the current frame's LSFs
|
|
1058 * for fixed_gain_smooth.
|
|
1059 * The specification has an incorrect formula: the reference decoder uses
|
|
1060 * qbar(n-1) rather than qbar(n) in section 6.1(4) equation 71. */
|
|
1061 ff_weighted_vector_sumf(p->lsf_avg, p->lsf_avg, p->lsf_q[3],
|
|
1062 0.84, 0.16, LP_FILTER_ORDER);
|
|
1063
|
|
1064 /* report how many samples we got */
|
|
1065 *data_size = AMR_BLOCK_SIZE * sizeof(float);
|
|
1066
|
|
1067 /* return the amount of bytes consumed if everything was OK */
|
|
1068 return frame_sizes_nb[p->cur_frame_mode] + 1; // +7 for rounding and +8 for TOC
|
|
1069 }
|
|
1070
|
|
1071
|
|
1072 AVCodec amrnb_decoder = {
|
|
1073 .name = "amrnb",
|
|
1074 .type = CODEC_TYPE_AUDIO,
|
|
1075 .id = CODEC_ID_AMR_NB,
|
|
1076 .priv_data_size = sizeof(AMRContext),
|
|
1077 .init = amrnb_decode_init,
|
|
1078 .decode = amrnb_decode_frame,
|
|
1079 .long_name = NULL_IF_CONFIG_SMALL("Adaptive Multi-Rate NarrowBand"),
|
|
1080 .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_FLT,SAMPLE_FMT_NONE},
|
|
1081 };
|