11123
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1 /*
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2 * Windows Media Audio Voice decoder.
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3 * Copyright (c) 2009 Ronald S. Bultje
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4 *
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5 * This file is part of FFmpeg.
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6 *
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7 * FFmpeg is free software; you can redistribute it and/or
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8 * modify it under the terms of the GNU Lesser General Public
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9 * License as published by the Free Software Foundation; either
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10 * version 2.1 of the License, or (at your option) any later version.
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11 *
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12 * FFmpeg is distributed in the hope that it will be useful,
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13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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15 * Lesser General Public License for more details.
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16 *
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17 * You should have received a copy of the GNU Lesser General Public
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18 * License along with FFmpeg; if not, write to the Free Software
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19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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20 */
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21
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22 /**
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23 * @file libavcodec/wmavoice.c
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24 * @brief Windows Media Audio Voice compatible decoder
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25 * @author Ronald S. Bultje <rsbultje@gmail.com>
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26 */
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27
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28 #include <math.h>
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29 #include "avcodec.h"
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30 #include "get_bits.h"
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31 #include "put_bits.h"
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32 #include "wmavoice_data.h"
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33 #include "celp_math.h"
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34 #include "celp_filters.h"
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35 #include "acelp_vectors.h"
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36 #include "acelp_filters.h"
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37 #include "lsp.h"
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38 #include "libavutil/lzo.h"
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39
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40 #define MAX_BLOCKS 8 ///< maximum number of blocks per frame
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41 #define MAX_LSPS 16 ///< maximum filter order
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42 #define MAX_FRAMES 3 ///< maximum number of frames per superframe
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43 #define MAX_FRAMESIZE 160 ///< maximum number of samples per frame
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44 #define MAX_SIGNAL_HISTORY 416 ///< maximum excitation signal history
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45 #define MAX_SFRAMESIZE (MAX_FRAMESIZE * MAX_FRAMES)
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46 ///< maximum number of samples per superframe
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47 #define SFRAME_CACHE_MAXSIZE 256 ///< maximum cache size for frame data that
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48 ///< was split over two packets
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49 #define VLC_NBITS 6 ///< number of bits to read per VLC iteration
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50
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51 /**
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52 * Frame type VLC coding.
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53 */
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54 static VLC frame_type_vlc;
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55
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56 /**
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57 * Adaptive codebook types.
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58 */
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59 enum {
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60 ACB_TYPE_NONE = 0, ///< no adaptive codebook (only hardcoded fixed)
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61 ACB_TYPE_ASYMMETRIC = 1, ///< adaptive codebook with per-frame pitch, which
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62 ///< we interpolate to get a per-sample pitch.
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63 ///< Signal is generated using an asymmetric sinc
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64 ///< window function
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65 ///< @note see #wmavoice_ipol1_coeffs
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66 ACB_TYPE_HAMMING = 2 ///< Per-block pitch with signal generation using
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67 ///< a Hamming sinc window function
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68 ///< @note see #wmavoice_ipol2_coeffs
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69 };
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70
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71 /**
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72 * Fixed codebook types.
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73 */
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74 enum {
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75 FCB_TYPE_SILENCE = 0, ///< comfort noise during silence
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76 ///< generated from a hardcoded (fixed) codebook
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77 ///< with per-frame (low) gain values
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78 FCB_TYPE_HARDCODED = 1, ///< hardcoded (fixed) codebook with per-block
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79 ///< gain values
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80 FCB_TYPE_AW_PULSES = 2, ///< Pitch-adaptive window (AW) pulse signals,
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81 ///< used in particular for low-bitrate streams
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82 FCB_TYPE_EXC_PULSES = 3, ///< Innovation (fixed) codebook pulse sets in
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83 ///< combinations of either single pulses or
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84 ///< pulse pairs
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85 };
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86
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87 /**
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88 * Description of frame types.
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89 */
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90 static const struct frame_type_desc {
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91 uint8_t n_blocks; ///< amount of blocks per frame (each block
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92 ///< (contains 160/#n_blocks samples)
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93 uint8_t log_n_blocks; ///< log2(#n_blocks)
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94 uint8_t acb_type; ///< Adaptive codebook type (ACB_TYPE_*)
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95 uint8_t fcb_type; ///< Fixed codebook type (FCB_TYPE_*)
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96 uint8_t dbl_pulses; ///< how many pulse vectors have pulse pairs
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97 ///< (rather than just one single pulse)
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98 ///< only if #fcb_type == #FCB_TYPE_EXC_PULSES
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99 uint16_t frame_size; ///< the amount of bits that make up the block
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100 ///< data (per frame)
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101 } frame_descs[17] = {
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102 { 1, 0, ACB_TYPE_NONE, FCB_TYPE_SILENCE, 0, 0 },
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103 { 2, 1, ACB_TYPE_NONE, FCB_TYPE_HARDCODED, 0, 28 },
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104 { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_AW_PULSES, 0, 46 },
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105 { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 2, 80 },
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106 { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 5, 104 },
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107 { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 0, 108 },
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108 { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 2, 132 },
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109 { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 5, 168 },
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110 { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0, 64 },
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111 { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2, 80 },
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112 { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5, 104 },
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113 { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0, 108 },
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114 { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2, 132 },
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115 { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5, 168 },
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116 { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0, 176 },
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117 { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2, 208 },
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118 { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5, 256 }
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119 };
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120
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121 /**
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122 * WMA Voice decoding context.
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123 */
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124 typedef struct {
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125 /**
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126 * @defgroup struct_global Global values
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127 * Global values, specified in the stream header / extradata or used
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128 * all over.
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129 * @{
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130 */
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131 GetBitContext gb; ///< packet bitreader. During decoder init,
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132 ///< it contains the extradata from the
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133 ///< demuxer. During decoding, it contains
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134 ///< packet data.
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135 int8_t vbm_tree[25]; ///< converts VLC codes to frame type
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136
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137 int spillover_bitsize; ///< number of bits used to specify
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138 ///< #spillover_nbits in the packet header
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139 ///< = ceil(log2(ctx->block_align << 3))
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140 int history_nsamples; ///< number of samples in history for signal
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141 ///< prediction (through ACB)
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142
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143 int do_apf; ///< whether to apply the averaged
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144 ///< projection filter (APF)
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145
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146 int lsps; ///< number of LSPs per frame [10 or 16]
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147 int lsp_q_mode; ///< defines quantizer defaults [0, 1]
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148 int lsp_def_mode; ///< defines different sets of LSP defaults
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149 ///< [0, 1]
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150 int frame_lsp_bitsize; ///< size (in bits) of LSPs, when encoded
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151 ///< per-frame (independent coding)
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152 int sframe_lsp_bitsize; ///< size (in bits) of LSPs, when encoded
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153 ///< per superframe (residual coding)
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154
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155 int min_pitch_val; ///< base value for pitch parsing code
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156 int max_pitch_val; ///< max value + 1 for pitch parsing
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157 int pitch_nbits; ///< number of bits used to specify the
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158 ///< pitch value in the frame header
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159 int block_pitch_nbits; ///< number of bits used to specify the
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160 ///< first block's pitch value
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161 int block_pitch_range; ///< range of the block pitch
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162 int block_delta_pitch_nbits; ///< number of bits used to specify the
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163 ///< delta pitch between this and the last
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164 ///< block's pitch value, used in all but
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165 ///< first block
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166 int block_delta_pitch_hrange; ///< 1/2 range of the delta (full range is
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167 ///< from -this to +this-1)
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168 uint16_t block_conv_table[4]; ///< boundaries for block pitch unit/scale
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169 ///< conversion
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170
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171 /**
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172 * @}
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173 * @defgroup struct_packet Packet values
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174 * Packet values, specified in the packet header or related to a packet.
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175 * A packet is considered to be a single unit of data provided to this
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176 * decoder by the demuxer.
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177 * @{
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178 */
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179 int spillover_nbits; ///< number of bits of the previous packet's
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180 ///< last superframe preceeding this
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181 ///< packet's first full superframe (useful
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182 ///< for re-synchronization also)
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183 int has_residual_lsps; ///< if set, superframes contain one set of
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184 ///< LSPs that cover all frames, encoded as
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185 ///< independent and residual LSPs; if not
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186 ///< set, each frame contains its own, fully
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187 ///< independent, LSPs
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188 int skip_bits_next; ///< number of bits to skip at the next call
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189 ///< to #wmavoice_decode_packet() (since
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190 ///< they're part of the previous superframe)
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191
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192 uint8_t sframe_cache[SFRAME_CACHE_MAXSIZE + FF_INPUT_BUFFER_PADDING_SIZE];
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193 ///< cache for superframe data split over
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194 ///< multiple packets
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195 int sframe_cache_size; ///< set to >0 if we have data from an
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196 ///< (incomplete) superframe from a previous
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197 ///< packet that spilled over in the current
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198 ///< packet; specifies the amount of bits in
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199 ///< #sframe_cache
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200 PutBitContext pb; ///< bitstream writer for #sframe_cache
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201
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202 /**
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203 * @}
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204 * @defgroup struct_frame Frame and superframe values
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205 * Superframe and frame data - these can change from frame to frame,
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206 * although some of them do in that case serve as a cache / history for
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207 * the next frame or superframe.
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208 * @{
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209 */
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210 double prev_lsps[MAX_LSPS]; ///< LSPs of the last frame of the previous
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211 ///< superframe
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212 int last_pitch_val; ///< pitch value of the previous frame
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213 int last_acb_type; ///< frame type [0-2] of the previous frame
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214 int pitch_diff_sh16; ///< ((cur_pitch_val - #last_pitch_val)
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215 ///< << 16) / #MAX_FRAMESIZE
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216 float silence_gain; ///< set for use in blocks if #ACB_TYPE_NONE
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217
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218 int aw_idx_is_ext; ///< whether the AW index was encoded in
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219 ///< 8 bits (instead of 6)
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220 int aw_pulse_range; ///< the range over which #aw_pulse_set1()
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221 ///< can apply the pulse, relative to the
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222 ///< value in aw_first_pulse_off. The exact
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223 ///< position of the first AW-pulse is within
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224 ///< [pulse_off, pulse_off + this], and
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225 ///< depends on bitstream values; [16 or 24]
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226 int aw_n_pulses[2]; ///< number of AW-pulses in each block; note
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227 ///< that this number can be negative (in
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228 ///< which case it basically means "zero")
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229 int aw_first_pulse_off[2]; ///< index of first sample to which to
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230 ///< apply AW-pulses, or -0xff if unset
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231 int aw_next_pulse_off_cache; ///< the position (relative to start of the
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232 ///< second block) at which pulses should
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233 ///< start to be positioned, serves as a
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234 ///< cache for pitch-adaptive window pulses
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235 ///< between blocks
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236
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237 int frame_cntr; ///< current frame index [0 - 0xFFFE]; is
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238 ///< only used for comfort noise in #pRNG()
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239 float gain_pred_err[6]; ///< cache for gain prediction
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240 float excitation_history[MAX_SIGNAL_HISTORY];
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241 ///< cache of the signal of previous
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242 ///< superframes, used as a history for
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243 ///< signal generation
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244 float synth_history[MAX_LSPS]; ///< see #excitation_history
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245 /**
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246 * @}
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247 */
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248 } WMAVoiceContext;
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249
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250 /**
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251 * Sets up the variable bit mode (VBM) tree from container extradata.
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252 * @param gb bit I/O context.
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253 * The bit context (s->gb) should be loaded with byte 23-46 of the
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254 * container extradata (i.e. the ones containing the VBM tree).
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255 * @param vbm_tree pointer to array to which the decoded VBM tree will be
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256 * written.
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257 * @return 0 on success, <0 on error.
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258 */
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259 static av_cold int decode_vbmtree(GetBitContext *gb, int8_t vbm_tree[25])
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260 {
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261 static const uint8_t bits[] = {
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262 2, 2, 2, 4, 4, 4,
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263 6, 6, 6, 8, 8, 8,
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264 10, 10, 10, 12, 12, 12,
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265 14, 14, 14, 14
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266 };
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267 static const uint16_t codes[] = {
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268 0x0000, 0x0001, 0x0002, // 00/01/10
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269 0x000c, 0x000d, 0x000e, // 11+00/01/10
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270 0x003c, 0x003d, 0x003e, // 1111+00/01/10
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271 0x00fc, 0x00fd, 0x00fe, // 111111+00/01/10
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272 0x03fc, 0x03fd, 0x03fe, // 11111111+00/01/10
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273 0x0ffc, 0x0ffd, 0x0ffe, // 1111111111+00/01/10
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274 0x3ffc, 0x3ffd, 0x3ffe, 0x3fff // 111111111111+xx
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275 };
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276 int cntr[8], n, res;
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277
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278 memset(vbm_tree, 0xff, sizeof(vbm_tree));
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279 memset(cntr, 0, sizeof(cntr));
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280 for (n = 0; n < 17; n++) {
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281 res = get_bits(gb, 3);
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282 if (cntr[res] > 3) // should be >= 3 + (res == 7))
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283 return -1;
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284 vbm_tree[res * 3 + cntr[res]++] = n;
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285 }
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286 INIT_VLC_STATIC(&frame_type_vlc, VLC_NBITS, sizeof(bits),
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287 bits, 1, 1, codes, 2, 2, 132);
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288 return 0;
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289 }
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290
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291 /**
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292 * Set up decoder with parameters from demuxer (extradata etc.).
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293 */
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294 static av_cold int wmavoice_decode_init(AVCodecContext *ctx)
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295 {
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296 int n, flags, pitch_range, lsp16_flag;
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297 WMAVoiceContext *s = ctx->priv_data;
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298
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299 /**
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300 * Extradata layout:
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301 * - byte 0-18: WMAPro-in-WMAVoice extradata (see wmaprodec.c),
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302 * - byte 19-22: flags field (annoyingly in LE; see below for known
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303 * values),
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304 * - byte 23-46: variable bitmode tree (really just 17 * 3 bits,
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305 * rest is 0).
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306 */
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307 if (ctx->extradata_size != 46) {
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308 av_log(ctx, AV_LOG_ERROR,
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309 "Invalid extradata size %d (should be 46)\n",
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310 ctx->extradata_size);
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311 return -1;
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312 }
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313 flags = AV_RL32(ctx->extradata + 18);
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314 s->spillover_bitsize = 3 + av_ceil_log2(ctx->block_align);
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315 s->do_apf = flags & 0x1;
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316 s->lsp_q_mode = !!(flags & 0x2000);
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317 s->lsp_def_mode = !!(flags & 0x4000);
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318 lsp16_flag = flags & 0x1000;
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319 if (lsp16_flag) {
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320 s->lsps = 16;
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321 s->frame_lsp_bitsize = 34;
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322 s->sframe_lsp_bitsize = 60;
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323 } else {
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324 s->lsps = 10;
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325 s->frame_lsp_bitsize = 24;
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326 s->sframe_lsp_bitsize = 48;
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327 }
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328 for (n = 0; n < s->lsps; n++)
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329 s->prev_lsps[n] = M_PI * (n + 1.0) / (s->lsps + 1.0);
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330
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331 init_get_bits(&s->gb, ctx->extradata + 22, (ctx->extradata_size - 22) << 3);
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332 if (decode_vbmtree(&s->gb, s->vbm_tree) < 0) {
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333 av_log(ctx, AV_LOG_ERROR, "Invalid VBM tree; broken extradata?\n");
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334 return -1;
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335 }
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336
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337 s->min_pitch_val = ((ctx->sample_rate << 8) / 400 + 50) >> 8;
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338 s->max_pitch_val = ((ctx->sample_rate << 8) * 37 / 2000 + 50) >> 8;
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339 pitch_range = s->max_pitch_val - s->min_pitch_val;
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340 s->pitch_nbits = av_ceil_log2(pitch_range);
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341 s->last_pitch_val = 40;
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342 s->last_acb_type = ACB_TYPE_NONE;
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343 s->history_nsamples = s->max_pitch_val + 8;
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344
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345 if (s->min_pitch_val < 1 || s->history_nsamples > MAX_SIGNAL_HISTORY) {
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346 int min_sr = ((((1 << 8) - 50) * 400) + 0xFF) >> 8,
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347 max_sr = ((((MAX_SIGNAL_HISTORY - 8) << 8) + 205) * 2000 / 37) >> 8;
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348
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349 av_log(ctx, AV_LOG_ERROR,
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350 "Unsupported samplerate %d (min=%d, max=%d)\n",
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351 ctx->sample_rate, min_sr, max_sr); // 322-22097 Hz
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352
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353 return -1;
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354 }
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355
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356 s->block_conv_table[0] = s->min_pitch_val;
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357 s->block_conv_table[1] = (pitch_range * 25) >> 6;
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358 s->block_conv_table[2] = (pitch_range * 44) >> 6;
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359 s->block_conv_table[3] = s->max_pitch_val - 1;
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360 s->block_delta_pitch_hrange = (pitch_range >> 3) & ~0xF;
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361 s->block_delta_pitch_nbits = 1 + av_ceil_log2(s->block_delta_pitch_hrange);
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362 s->block_pitch_range = s->block_conv_table[2] +
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363 s->block_conv_table[3] + 1 +
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364 2 * (s->block_conv_table[1] - 2 * s->min_pitch_val);
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365 s->block_pitch_nbits = av_ceil_log2(s->block_pitch_range);
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366
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367 ctx->sample_fmt = SAMPLE_FMT_FLT;
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368
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369 return 0;
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370 }
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371
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372 /**
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373 * Dequantize LSPs
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374 * @param lsps output pointer to the array that will hold the LSPs
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375 * @param num number of LSPs to be dequantized
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376 * @param values quantized values, contains n_stages values
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377 * @param sizes range (i.e. max value) of each quantized value
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378 * @param n_stages number of dequantization runs
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379 * @param table dequantization table to be used
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380 * @param mul_q LSF multiplier
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381 * @param base_q base (lowest) LSF values
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382 */
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383 static void dequant_lsps(double *lsps, int num,
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384 const uint16_t *values,
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385 const uint16_t *sizes,
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386 int n_stages, const uint8_t *table,
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387 const double *mul_q,
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388 const double *base_q)
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389 {
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390 int n, m;
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391
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392 memset(lsps, 0, num * sizeof(*lsps));
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393 for (n = 0; n < n_stages; n++) {
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394 const uint8_t *t_off = &table[values[n] * num];
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395 double base = base_q[n], mul = mul_q[n];
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396
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397 for (m = 0; m < num; m++)
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398 lsps[m] += base + mul * t_off[m];
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399
|
|
400 table += sizes[n] * num;
|
|
401 }
|
|
402 }
|
|
403
|
|
404 /**
|
|
405 * @defgroup lsp_dequant LSP dequantization routines
|
|
406 * LSP dequantization routines, for 10/16LSPs and independent/residual coding.
|
|
407 * @note we assume enough bits are available, caller should check.
|
|
408 * lsp10i() consumes 24 bits; lsp10r() consumes an additional 24 bits;
|
|
409 * lsp16i() consumes 34 bits; lsp16r() consumes an additional 26 bits.
|
|
410 * @{
|
|
411 */
|
|
412 /**
|
|
413 * Parse 10 independently-coded LSPs.
|
|
414 */
|
|
415 static void dequant_lsp10i(GetBitContext *gb, double *lsps)
|
|
416 {
|
|
417 static const uint16_t vec_sizes[4] = { 256, 64, 32, 32 };
|
|
418 static const double mul_lsf[4] = {
|
|
419 5.2187144800e-3, 1.4626986422e-3,
|
|
420 9.6179549166e-4, 1.1325736225e-3
|
|
421 };
|
|
422 static const double base_lsf[4] = {
|
|
423 M_PI * -2.15522e-1, M_PI * -6.1646e-2,
|
|
424 M_PI * -3.3486e-2, M_PI * -5.7408e-2
|
|
425 };
|
|
426 uint16_t v[4];
|
|
427
|
|
428 v[0] = get_bits(gb, 8);
|
|
429 v[1] = get_bits(gb, 6);
|
|
430 v[2] = get_bits(gb, 5);
|
|
431 v[3] = get_bits(gb, 5);
|
|
432
|
|
433 dequant_lsps(lsps, 10, v, vec_sizes, 4, wmavoice_dq_lsp10i,
|
|
434 mul_lsf, base_lsf);
|
|
435 }
|
|
436
|
|
437 /**
|
|
438 * Parse 10 independently-coded LSPs, and then derive the tables to
|
|
439 * generate LSPs for the other frames from them (residual coding).
|
|
440 */
|
|
441 static void dequant_lsp10r(GetBitContext *gb,
|
|
442 double *i_lsps, const double *old,
|
|
443 double *a1, double *a2, int q_mode)
|
|
444 {
|
|
445 static const uint16_t vec_sizes[3] = { 128, 64, 64 };
|
|
446 static const double mul_lsf[3] = {
|
|
447 2.5807601174e-3, 1.2354460219e-3, 1.1763821673e-3
|
|
448 };
|
|
449 static const double base_lsf[3] = {
|
|
450 M_PI * -1.07448e-1, M_PI * -5.2706e-2, M_PI * -5.1634e-2
|
|
451 };
|
|
452 const float (*ipol_tab)[2][10] = q_mode ?
|
|
453 wmavoice_lsp10_intercoeff_b : wmavoice_lsp10_intercoeff_a;
|
|
454 uint16_t interpol, v[3];
|
|
455 int n;
|
|
456
|
|
457 dequant_lsp10i(gb, i_lsps);
|
|
458
|
|
459 interpol = get_bits(gb, 5);
|
|
460 v[0] = get_bits(gb, 7);
|
|
461 v[1] = get_bits(gb, 6);
|
|
462 v[2] = get_bits(gb, 6);
|
|
463
|
|
464 for (n = 0; n < 10; n++) {
|
|
465 double delta = old[n] - i_lsps[n];
|
|
466 a1[n] = ipol_tab[interpol][0][n] * delta + i_lsps[n];
|
|
467 a1[10 + n] = ipol_tab[interpol][1][n] * delta + i_lsps[n];
|
|
468 }
|
|
469
|
|
470 dequant_lsps(a2, 20, v, vec_sizes, 3, wmavoice_dq_lsp10r,
|
|
471 mul_lsf, base_lsf);
|
|
472 }
|
|
473
|
|
474 /**
|
|
475 * Parse 16 independently-coded LSPs.
|
|
476 */
|
|
477 static void dequant_lsp16i(GetBitContext *gb, double *lsps)
|
|
478 {
|
|
479 static const uint16_t vec_sizes[5] = { 256, 64, 128, 64, 128 };
|
|
480 static const double mul_lsf[5] = {
|
|
481 3.3439586280e-3, 6.9908173703e-4,
|
|
482 3.3216608306e-3, 1.0334960326e-3,
|
|
483 3.1899104283e-3
|
|
484 };
|
|
485 static const double base_lsf[5] = {
|
|
486 M_PI * -1.27576e-1, M_PI * -2.4292e-2,
|
|
487 M_PI * -1.28094e-1, M_PI * -3.2128e-2,
|
|
488 M_PI * -1.29816e-1
|
|
489 };
|
|
490 uint16_t v[5];
|
|
491
|
|
492 v[0] = get_bits(gb, 8);
|
|
493 v[1] = get_bits(gb, 6);
|
|
494 v[2] = get_bits(gb, 7);
|
|
495 v[3] = get_bits(gb, 6);
|
|
496 v[4] = get_bits(gb, 7);
|
|
497
|
|
498 dequant_lsps( lsps, 5, v, vec_sizes, 2,
|
|
499 wmavoice_dq_lsp16i1, mul_lsf, base_lsf);
|
|
500 dequant_lsps(&lsps[5], 5, &v[2], &vec_sizes[2], 2,
|
|
501 wmavoice_dq_lsp16i2, &mul_lsf[2], &base_lsf[2]);
|
|
502 dequant_lsps(&lsps[10], 6, &v[4], &vec_sizes[4], 1,
|
|
503 wmavoice_dq_lsp16i3, &mul_lsf[4], &base_lsf[4]);
|
|
504 }
|
|
505
|
|
506 /**
|
|
507 * Parse 16 independently-coded LSPs, and then derive the tables to
|
|
508 * generate LSPs for the other frames from them (residual coding).
|
|
509 */
|
|
510 static void dequant_lsp16r(GetBitContext *gb,
|
|
511 double *i_lsps, const double *old,
|
|
512 double *a1, double *a2, int q_mode)
|
|
513 {
|
|
514 static const uint16_t vec_sizes[3] = { 128, 128, 128 };
|
|
515 static const double mul_lsf[3] = {
|
|
516 1.2232979501e-3, 1.4062241527e-3, 1.6114744851e-3
|
|
517 };
|
|
518 static const double base_lsf[3] = {
|
|
519 M_PI * -5.5830e-2, M_PI * -5.2908e-2, M_PI * -5.4776e-2
|
|
520 };
|
|
521 const float (*ipol_tab)[2][16] = q_mode ?
|
|
522 wmavoice_lsp16_intercoeff_b : wmavoice_lsp16_intercoeff_a;
|
|
523 uint16_t interpol, v[3];
|
|
524 int n;
|
|
525
|
|
526 dequant_lsp16i(gb, i_lsps);
|
|
527
|
|
528 interpol = get_bits(gb, 5);
|
|
529 v[0] = get_bits(gb, 7);
|
|
530 v[1] = get_bits(gb, 7);
|
|
531 v[2] = get_bits(gb, 7);
|
|
532
|
|
533 for (n = 0; n < 16; n++) {
|
|
534 double delta = old[n] - i_lsps[n];
|
|
535 a1[n] = ipol_tab[interpol][0][n] * delta + i_lsps[n];
|
|
536 a1[16 + n] = ipol_tab[interpol][1][n] * delta + i_lsps[n];
|
|
537 }
|
|
538
|
|
539 dequant_lsps( a2, 10, v, vec_sizes, 1,
|
|
540 wmavoice_dq_lsp16r1, mul_lsf, base_lsf);
|
|
541 dequant_lsps(&a2[10], 10, &v[1], &vec_sizes[1], 1,
|
|
542 wmavoice_dq_lsp16r2, &mul_lsf[1], &base_lsf[1]);
|
|
543 dequant_lsps(&a2[20], 12, &v[2], &vec_sizes[2], 1,
|
|
544 wmavoice_dq_lsp16r3, &mul_lsf[2], &base_lsf[2]);
|
|
545 }
|
|
546
|
|
547 /**
|
|
548 * @}
|
|
549 * @defgroup aw Pitch-adaptive window coding functions
|
|
550 * The next few functions are for pitch-adaptive window coding.
|
|
551 * @{
|
|
552 */
|
|
553 /**
|
|
554 * Parse the offset of the first pitch-adaptive window pulses, and
|
|
555 * the distribution of pulses between the two blocks in this frame.
|
|
556 * @param s WMA Voice decoding context private data
|
|
557 * @param gb bit I/O context
|
|
558 * @param pitch pitch for each block in this frame
|
|
559 */
|
|
560 static void aw_parse_coords(WMAVoiceContext *s, GetBitContext *gb,
|
|
561 const int *pitch)
|
|
562 {
|
|
563 static const int16_t start_offset[94] = {
|
|
564 -11, -9, -7, -5, -3, -1, 1, 3, 5, 7, 9, 11,
|
|
565 13, 15, 18, 17, 19, 20, 21, 22, 23, 24, 25, 26,
|
|
566 27, 28, 29, 30, 31, 32, 33, 35, 37, 39, 41, 43,
|
|
567 45, 47, 49, 51, 53, 55, 57, 59, 61, 63, 65, 67,
|
|
568 69, 71, 73, 75, 77, 79, 81, 83, 85, 87, 89, 91,
|
|
569 93, 95, 97, 99, 101, 103, 105, 107, 109, 111, 113, 115,
|
|
570 117, 119, 121, 123, 125, 127, 129, 131, 133, 135, 137, 139,
|
|
571 141, 143, 145, 147, 149, 151, 153, 155, 157, 159
|
|
572 };
|
|
573 int bits, offset;
|
|
574
|
|
575 /* position of pulse */
|
|
576 s->aw_idx_is_ext = 0;
|
|
577 if ((bits = get_bits(gb, 6)) >= 54) {
|
|
578 s->aw_idx_is_ext = 1;
|
|
579 bits += (bits - 54) * 3 + get_bits(gb, 2);
|
|
580 }
|
|
581
|
|
582 /* for a repeated pulse at pulse_off with a pitch_lag of pitch[], count
|
|
583 * the distribution of the pulses in each block contained in this frame. */
|
|
584 s->aw_pulse_range = FFMIN(pitch[0], pitch[1]) > 32 ? 24 : 16;
|
|
585 for (offset = start_offset[bits]; offset < 0; offset += pitch[0]) ;
|
|
586 s->aw_n_pulses[0] = (pitch[0] - 1 + MAX_FRAMESIZE / 2 - offset) / pitch[0];
|
|
587 s->aw_first_pulse_off[0] = offset - s->aw_pulse_range / 2;
|
|
588 offset += s->aw_n_pulses[0] * pitch[0];
|
|
589 s->aw_n_pulses[1] = (pitch[1] - 1 + MAX_FRAMESIZE - offset) / pitch[1];
|
|
590 s->aw_first_pulse_off[1] = offset - (MAX_FRAMESIZE + s->aw_pulse_range) / 2;
|
|
591
|
|
592 /* if continuing from a position before the block, reset position to
|
|
593 * start of block (when corrected for the range over which it can be
|
|
594 * spread in aw_pulse_set1()). */
|
|
595 if (start_offset[bits] < MAX_FRAMESIZE / 2) {
|
|
596 while (s->aw_first_pulse_off[1] - pitch[1] + s->aw_pulse_range > 0)
|
|
597 s->aw_first_pulse_off[1] -= pitch[1];
|
|
598 if (start_offset[bits] < 0)
|
|
599 while (s->aw_first_pulse_off[0] - pitch[0] + s->aw_pulse_range > 0)
|
|
600 s->aw_first_pulse_off[0] -= pitch[0];
|
|
601 }
|
|
602 }
|
|
603
|
|
604 /**
|
|
605 * Apply second set of pitch-adaptive window pulses.
|
|
606 * @param s WMA Voice decoding context private data
|
|
607 * @param gb bit I/O context
|
|
608 * @param block_idx block index in frame [0, 1]
|
|
609 * @param fcb structure containing fixed codebook vector info
|
|
610 */
|
|
611 static void aw_pulse_set2(WMAVoiceContext *s, GetBitContext *gb,
|
|
612 int block_idx, AMRFixed *fcb)
|
|
613 {
|
|
614 uint16_t use_mask[7]; // only 5 are used, rest is padding
|
|
615 /* in this function, idx is the index in the 80-bit (+ padding) use_mask
|
|
616 * bit-array. Since use_mask consists of 16-bit values, the lower 4 bits
|
|
617 * of idx are the position of the bit within a particular item in the
|
|
618 * array (0 being the most significant bit, and 15 being the least
|
|
619 * significant bit), and the remainder (>> 4) is the index in the
|
|
620 * use_mask[]-array. This is faster and uses less memory than using a
|
|
621 * 80-byte/80-int array. */
|
|
622 int pulse_off = s->aw_first_pulse_off[block_idx],
|
|
623 pulse_start, n, idx, range, aidx, start_off = 0;
|
|
624
|
|
625 /* set offset of first pulse to within this block */
|
|
626 if (s->aw_n_pulses[block_idx] > 0)
|
|
627 while (pulse_off + s->aw_pulse_range < 1)
|
|
628 pulse_off += fcb->pitch_lag;
|
|
629
|
|
630 /* find range per pulse */
|
|
631 if (s->aw_n_pulses[0] > 0) {
|
|
632 if (block_idx == 0) {
|
|
633 range = 32;
|
|
634 } else /* block_idx = 1 */ {
|
|
635 range = 8;
|
|
636 if (s->aw_n_pulses[block_idx] > 0)
|
|
637 pulse_off = s->aw_next_pulse_off_cache;
|
|
638 }
|
|
639 } else
|
|
640 range = 16;
|
|
641 pulse_start = s->aw_n_pulses[block_idx] > 0 ? pulse_off - range / 2 : 0;
|
|
642
|
|
643 /* aw_pulse_set1() already applies pulses around pulse_off (to be exactly,
|
|
644 * in the range of [pulse_off, pulse_off + s->aw_pulse_range], and thus
|
|
645 * we exclude that range from being pulsed again in this function. */
|
|
646 memset( use_mask, -1, 5 * sizeof(use_mask[0]));
|
|
647 memset(&use_mask[5], 0, 2 * sizeof(use_mask[0]));
|
|
648 if (s->aw_n_pulses[block_idx] > 0)
|
|
649 for (idx = pulse_off; idx < MAX_FRAMESIZE / 2; idx += fcb->pitch_lag) {
|
|
650 int excl_range = s->aw_pulse_range; // always 16 or 24
|
|
651 uint16_t *use_mask_ptr = &use_mask[idx >> 4];
|
|
652 int first_sh = 16 - (idx & 15);
|
|
653 *use_mask_ptr++ &= 0xFFFF << first_sh;
|
|
654 excl_range -= first_sh;
|
|
655 if (excl_range >= 16) {
|
|
656 *use_mask_ptr++ = 0;
|
|
657 *use_mask_ptr &= 0xFFFF >> (excl_range - 16);
|
|
658 } else
|
|
659 *use_mask_ptr &= 0xFFFF >> excl_range;
|
|
660 }
|
|
661
|
|
662 /* find the 'aidx'th offset that is not excluded */
|
|
663 aidx = get_bits(gb, s->aw_n_pulses[0] > 0 ? 5 - 2 * block_idx : 4);
|
|
664 for (n = 0; n <= aidx; pulse_start++) {
|
|
665 for (idx = pulse_start; idx < 0; idx += fcb->pitch_lag) ;
|
|
666 if (idx >= MAX_FRAMESIZE / 2) { // find from zero
|
|
667 if (use_mask[0]) idx = 0x0F;
|
|
668 else if (use_mask[1]) idx = 0x1F;
|
|
669 else if (use_mask[2]) idx = 0x2F;
|
|
670 else if (use_mask[3]) idx = 0x3F;
|
|
671 else if (use_mask[4]) idx = 0x4F;
|
|
672 else return;
|
|
673 idx -= av_log2_16bit(use_mask[idx >> 4]);
|
|
674 }
|
|
675 if (use_mask[idx >> 4] & (0x8000 >> (idx & 15))) {
|
|
676 use_mask[idx >> 4] &= ~(0x8000 >> (idx & 15));
|
|
677 n++;
|
|
678 start_off = idx;
|
|
679 }
|
|
680 }
|
|
681
|
|
682 fcb->x[fcb->n] = start_off;
|
|
683 fcb->y[fcb->n] = get_bits1(gb) ? -1.0 : 1.0;
|
|
684 fcb->n++;
|
|
685
|
|
686 /* set offset for next block, relative to start of that block */
|
|
687 n = (MAX_FRAMESIZE / 2 - start_off) % fcb->pitch_lag;
|
|
688 s->aw_next_pulse_off_cache = n ? fcb->pitch_lag - n : 0;
|
|
689 }
|
|
690
|
|
691 /**
|
|
692 * Apply first set of pitch-adaptive window pulses.
|
|
693 * @param s WMA Voice decoding context private data
|
|
694 * @param gb bit I/O context
|
|
695 * @param block_idx block index in frame [0, 1]
|
|
696 * @param fcb storage location for fixed codebook pulse info
|
|
697 */
|
|
698 static void aw_pulse_set1(WMAVoiceContext *s, GetBitContext *gb,
|
|
699 int block_idx, AMRFixed *fcb)
|
|
700 {
|
|
701 int val = get_bits(gb, 12 - 2 * (s->aw_idx_is_ext && !block_idx));
|
|
702 float v;
|
|
703
|
|
704 if (s->aw_n_pulses[block_idx] > 0) {
|
|
705 int n, v_mask, i_mask, sh, n_pulses;
|
|
706
|
|
707 if (s->aw_pulse_range == 24) { // 3 pulses, 1:sign + 3:index each
|
|
708 n_pulses = 3;
|
|
709 v_mask = 8;
|
|
710 i_mask = 7;
|
|
711 sh = 4;
|
|
712 } else { // 4 pulses, 1:sign + 2:index each
|
|
713 n_pulses = 4;
|
|
714 v_mask = 4;
|
|
715 i_mask = 3;
|
|
716 sh = 3;
|
|
717 }
|
|
718
|
|
719 for (n = n_pulses - 1; n >= 0; n--, val >>= sh) {
|
|
720 fcb->y[fcb->n] = (val & v_mask) ? -1.0 : 1.0;
|
|
721 fcb->x[fcb->n] = (val & i_mask) * n_pulses + n +
|
|
722 s->aw_first_pulse_off[block_idx];
|
|
723 while (fcb->x[fcb->n] < 0)
|
|
724 fcb->x[fcb->n] += fcb->pitch_lag;
|
|
725 if (fcb->x[fcb->n] < MAX_FRAMESIZE / 2)
|
|
726 fcb->n++;
|
|
727 }
|
|
728 } else {
|
|
729 int num2 = (val & 0x1FF) >> 1, delta, idx;
|
|
730
|
|
731 if (num2 < 1 * 79) { delta = 1; idx = num2 + 1; }
|
|
732 else if (num2 < 2 * 78) { delta = 3; idx = num2 + 1 - 1 * 77; }
|
|
733 else if (num2 < 3 * 77) { delta = 5; idx = num2 + 1 - 2 * 76; }
|
|
734 else { delta = 7; idx = num2 + 1 - 3 * 75; }
|
|
735 v = (val & 0x200) ? -1.0 : 1.0;
|
|
736
|
|
737 fcb->no_repeat_mask |= 3 << fcb->n;
|
|
738 fcb->x[fcb->n] = idx - delta;
|
|
739 fcb->y[fcb->n] = v;
|
|
740 fcb->x[fcb->n + 1] = idx;
|
|
741 fcb->y[fcb->n + 1] = (val & 1) ? -v : v;
|
|
742 fcb->n += 2;
|
|
743 }
|
|
744 }
|
|
745
|
|
746 /**
|
|
747 * @}
|
|
748 *
|
|
749 * Generate a random number from frame_cntr and block_idx, which will lief
|
|
750 * in the range [0, 1000 - block_size] (so it can be used as an index in a
|
|
751 * table of size 1000 of which you want to read block_size entries).
|
|
752 *
|
|
753 * @param frame_cntr current frame number
|
|
754 * @param block_num current block index
|
|
755 * @param block_size amount of entries we want to read from a table
|
|
756 * that has 1000 entries
|
|
757 * @returns a (non-)random number in the [0, 1000 - block_size] range.
|
|
758 */
|
|
759 static int pRNG(int frame_cntr, int block_num, int block_size)
|
|
760 {
|
|
761 /* array to simplify the calculation of z:
|
|
762 * y = (x % 9) * 5 + 6;
|
|
763 * z = (49995 * x) / y;
|
|
764 * Since y only has 9 values, we can remove the division by using a
|
|
765 * LUT and using FASTDIV-style divisions. For each of the 9 values
|
|
766 * of y, we can rewrite z as:
|
|
767 * z = x * (49995 / y) + x * ((49995 % y) / y)
|
|
768 * In this table, each col represents one possible value of y, the
|
|
769 * first number is 49995 / y, and the second is the FASTDIV variant
|
|
770 * of 49995 % y / y. */
|
|
771 static const unsigned int div_tbl[9][2] = {
|
|
772 { 8332, 3 * 715827883U }, // y = 6
|
|
773 { 4545, 0 * 390451573U }, // y = 11
|
|
774 { 3124, 11 * 268435456U }, // y = 16
|
|
775 { 2380, 15 * 204522253U }, // y = 21
|
|
776 { 1922, 23 * 165191050U }, // y = 26
|
|
777 { 1612, 23 * 138547333U }, // y = 31
|
|
778 { 1388, 27 * 119304648U }, // y = 36
|
|
779 { 1219, 16 * 104755300U }, // y = 41
|
|
780 { 1086, 39 * 93368855U } // y = 46
|
|
781 };
|
|
782 unsigned int z, y, x = MUL16(block_num, 1877) + frame_cntr;
|
|
783 if (x >= 0xFFFF) x -= 0xFFFF; // max value of x is 8*1877+0xFFFE=0x13AA6,
|
|
784 // so this is effectively a modulo (%)
|
|
785 y = x - 9 * MULH(477218589, x); // x % 9
|
|
786 z = (uint16_t) (x * div_tbl[y][0] + UMULH(x, div_tbl[y][1]));
|
|
787 // z = x * 49995 / (y * 5 + 6)
|
|
788 return z % (1000 - block_size);
|
|
789 }
|
|
790
|
|
791 /**
|
|
792 * Parse hardcoded signal for a single block.
|
|
793 * @note see #synth_block().
|
|
794 */
|
|
795 static void synth_block_hardcoded(WMAVoiceContext *s, GetBitContext *gb,
|
|
796 int block_idx, int size,
|
|
797 const struct frame_type_desc *frame_desc,
|
|
798 float *excitation)
|
|
799 {
|
|
800 float gain;
|
|
801 int n, r_idx;
|
|
802
|
|
803 assert(size <= MAX_FRAMESIZE);
|
|
804
|
|
805 /* Set the offset from which we start reading wmavoice_std_codebook */
|
|
806 if (frame_desc->fcb_type == FCB_TYPE_SILENCE) {
|
|
807 r_idx = pRNG(s->frame_cntr, block_idx, size);
|
|
808 gain = s->silence_gain;
|
|
809 } else /* FCB_TYPE_HARDCODED */ {
|
|
810 r_idx = get_bits(gb, 8);
|
|
811 gain = wmavoice_gain_universal[get_bits(gb, 6)];
|
|
812 }
|
|
813
|
|
814 /* Clear gain prediction parameters */
|
|
815 memset(s->gain_pred_err, 0, sizeof(s->gain_pred_err));
|
|
816
|
|
817 /* Apply gain to hardcoded codebook and use that as excitation signal */
|
|
818 for (n = 0; n < size; n++)
|
|
819 excitation[n] = wmavoice_std_codebook[r_idx + n] * gain;
|
|
820 }
|
|
821
|
|
822 /**
|
|
823 * Parse FCB/ACB signal for a single block.
|
|
824 * @note see #synth_block().
|
|
825 */
|
|
826 static void synth_block_fcb_acb(WMAVoiceContext *s, GetBitContext *gb,
|
|
827 int block_idx, int size,
|
|
828 int block_pitch_sh2,
|
|
829 const struct frame_type_desc *frame_desc,
|
|
830 float *excitation)
|
|
831 {
|
|
832 static const float gain_coeff[6] = {
|
|
833 0.8169, -0.06545, 0.1726, 0.0185, -0.0359, 0.0458
|
|
834 };
|
|
835 float pulses[MAX_FRAMESIZE / 2], pred_err, acb_gain, fcb_gain;
|
|
836 int n, idx, gain_weight;
|
|
837 AMRFixed fcb;
|
|
838
|
|
839 assert(size <= MAX_FRAMESIZE / 2);
|
|
840 memset(pulses, 0, sizeof(*pulses) * size);
|
|
841
|
|
842 fcb.pitch_lag = block_pitch_sh2 >> 2;
|
|
843 fcb.pitch_fac = 1.0;
|
|
844 fcb.no_repeat_mask = 0;
|
|
845 fcb.n = 0;
|
|
846
|
|
847 /* For the other frame types, this is where we apply the innovation
|
|
848 * (fixed) codebook pulses of the speech signal. */
|
|
849 if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) {
|
|
850 aw_pulse_set1(s, gb, block_idx, &fcb);
|
|
851 aw_pulse_set2(s, gb, block_idx, &fcb);
|
|
852 } else /* FCB_TYPE_EXC_PULSES */ {
|
|
853 int offset_nbits = 5 - frame_desc->log_n_blocks;
|
|
854
|
|
855 fcb.no_repeat_mask = -1;
|
|
856 /* similar to ff_decode_10_pulses_35bits(), but with single pulses
|
|
857 * (instead of double) for a subset of pulses */
|
|
858 for (n = 0; n < 5; n++) {
|
|
859 float sign;
|
|
860 int pos1, pos2;
|
|
861
|
|
862 sign = get_bits1(gb) ? 1.0 : -1.0;
|
|
863 pos1 = get_bits(gb, offset_nbits);
|
|
864 fcb.x[fcb.n] = n + 5 * pos1;
|
|
865 fcb.y[fcb.n++] = sign;
|
|
866 if (n < frame_desc->dbl_pulses) {
|
|
867 pos2 = get_bits(gb, offset_nbits);
|
|
868 fcb.x[fcb.n] = n + 5 * pos2;
|
|
869 fcb.y[fcb.n++] = (pos1 < pos2) ? -sign : sign;
|
|
870 }
|
|
871 }
|
|
872 }
|
|
873 ff_set_fixed_vector(pulses, &fcb, 1.0, size);
|
|
874
|
|
875 /* Calculate gain for adaptive & fixed codebook signal.
|
|
876 * see ff_amr_set_fixed_gain(). */
|
|
877 idx = get_bits(gb, 7);
|
|
878 fcb_gain = expf(ff_dot_productf(s->gain_pred_err, gain_coeff, 6) -
|
|
879 5.2409161640 + wmavoice_gain_codebook_fcb[idx]);
|
|
880 acb_gain = wmavoice_gain_codebook_acb[idx];
|
|
881 pred_err = av_clipf(wmavoice_gain_codebook_fcb[idx],
|
|
882 -2.9957322736 /* log(0.05) */,
|
|
883 1.6094379124 /* log(5.0) */);
|
|
884
|
|
885 gain_weight = 8 >> frame_desc->log_n_blocks;
|
|
886 memmove(&s->gain_pred_err[gain_weight], s->gain_pred_err,
|
|
887 sizeof(*s->gain_pred_err) * (6 - gain_weight));
|
|
888 for (n = 0; n < gain_weight; n++)
|
|
889 s->gain_pred_err[n] = pred_err;
|
|
890
|
|
891 /* Calculation of adaptive codebook */
|
|
892 if (frame_desc->acb_type == ACB_TYPE_ASYMMETRIC) {
|
|
893 int len;
|
|
894 for (n = 0; n < size; n += len) {
|
|
895 int next_idx_sh16;
|
|
896 int abs_idx = block_idx * size + n;
|
|
897 int pitch_sh16 = (s->last_pitch_val << 16) +
|
|
898 s->pitch_diff_sh16 * abs_idx;
|
|
899 int pitch = (pitch_sh16 + 0x6FFF) >> 16;
|
|
900 int idx_sh16 = ((pitch << 16) - pitch_sh16) * 8 + 0x58000;
|
|
901 idx = idx_sh16 >> 16;
|
|
902 if (s->pitch_diff_sh16) {
|
|
903 if (s->pitch_diff_sh16 > 0) {
|
|
904 next_idx_sh16 = (idx_sh16) &~ 0xFFFF;
|
|
905 } else
|
|
906 next_idx_sh16 = (idx_sh16 + 0x10000) &~ 0xFFFF;
|
|
907 len = av_clip((idx_sh16 - next_idx_sh16) / s->pitch_diff_sh16 / 8,
|
|
908 1, size - n);
|
|
909 } else
|
|
910 len = size;
|
|
911
|
|
912 ff_acelp_interpolatef(&excitation[n], &excitation[n - pitch],
|
|
913 wmavoice_ipol1_coeffs, 17,
|
|
914 idx, 9, len);
|
|
915 }
|
|
916 } else /* ACB_TYPE_HAMMING */ {
|
|
917 int block_pitch = block_pitch_sh2 >> 2;
|
|
918 idx = block_pitch_sh2 & 3;
|
|
919 if (idx) {
|
|
920 ff_acelp_interpolatef(excitation, &excitation[-block_pitch],
|
|
921 wmavoice_ipol2_coeffs, 4,
|
|
922 idx, 8, size);
|
|
923 } else
|
|
924 av_memcpy_backptr(excitation, sizeof(float) * block_pitch,
|
|
925 sizeof(float) * size);
|
|
926 }
|
|
927
|
|
928 /* Interpolate ACB/FCB and use as excitation signal */
|
|
929 ff_weighted_vector_sumf(excitation, excitation, pulses,
|
|
930 acb_gain, fcb_gain, size);
|
|
931 }
|
|
932
|
|
933 /**
|
|
934 * Parse data in a single block.
|
|
935 * @note we assume enough bits are available, caller should check.
|
|
936 *
|
|
937 * @param s WMA Voice decoding context private data
|
|
938 * @param gb bit I/O context
|
|
939 * @param block_idx index of the to-be-read block
|
|
940 * @param size amount of samples to be read in this block
|
|
941 * @param block_pitch_sh2 pitch for this block << 2
|
|
942 * @param lsps LSPs for (the end of) this frame
|
|
943 * @param prev_lsps LSPs for the last frame
|
|
944 * @param frame_desc frame type descriptor
|
|
945 * @param excitation target memory for the ACB+FCB interpolated signal
|
|
946 * @param synth target memory for the speech synthesis filter output
|
|
947 * @return 0 on success, <0 on error.
|
|
948 */
|
|
949 static void synth_block(WMAVoiceContext *s, GetBitContext *gb,
|
|
950 int block_idx, int size,
|
|
951 int block_pitch_sh2,
|
|
952 const double *lsps, const double *prev_lsps,
|
|
953 const struct frame_type_desc *frame_desc,
|
|
954 float *excitation, float *synth)
|
|
955 {
|
|
956 double i_lsps[MAX_LSPS];
|
|
957 float lpcs[MAX_LSPS];
|
|
958 float fac;
|
|
959 int n;
|
|
960
|
|
961 if (frame_desc->acb_type == ACB_TYPE_NONE)
|
|
962 synth_block_hardcoded(s, gb, block_idx, size, frame_desc, excitation);
|
|
963 else
|
|
964 synth_block_fcb_acb(s, gb, block_idx, size, block_pitch_sh2,
|
|
965 frame_desc, excitation);
|
|
966
|
|
967 /* convert interpolated LSPs to LPCs */
|
|
968 fac = (block_idx + 0.5) / frame_desc->n_blocks;
|
|
969 for (n = 0; n < s->lsps; n++) // LSF -> LSP
|
|
970 i_lsps[n] = cos(prev_lsps[n] + fac * (lsps[n] - prev_lsps[n]));
|
|
971 ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1);
|
|
972
|
|
973 /* Speech synthesis */
|
|
974 ff_celp_lp_synthesis_filterf(synth, lpcs, excitation, size, s->lsps);
|
|
975 }
|
|
976
|
|
977 /**
|
|
978 * Synthesize output samples for a single frame.
|
|
979 * @note we assume enough bits are available, caller should check.
|
|
980 *
|
|
981 * @param ctx WMA Voice decoder context
|
|
982 * @param gb bit I/O context (s->gb or one for cross-packet superframes)
|
|
983 * @param samples pointer to output sample buffer, has space for at least 160
|
|
984 * samples
|
|
985 * @param lsps LSP array
|
|
986 * @param prev_lsps array of previous frame's LSPs
|
|
987 * @param excitation target buffer for excitation signal
|
|
988 * @param synth target buffer for synthesized speech data
|
|
989 * @return 0 on success, <0 on error.
|
|
990 */
|
|
991 static int synth_frame(AVCodecContext *ctx, GetBitContext *gb,
|
|
992 float *samples,
|
|
993 const double *lsps, const double *prev_lsps,
|
|
994 float *excitation, float *synth)
|
|
995 {
|
|
996 WMAVoiceContext *s = ctx->priv_data;
|
|
997 int n, n_blocks_x2, log_n_blocks_x2, cur_pitch_val;
|
|
998 int pitch[MAX_BLOCKS], last_block_pitch;
|
|
999
|
|
1000 /* Parse frame type ("frame header"), see frame_descs */
|
|
1001 int bd_idx = s->vbm_tree[get_vlc2(gb, frame_type_vlc.table, 6, 3)],
|
|
1002 block_nsamples = MAX_FRAMESIZE / frame_descs[bd_idx].n_blocks;
|
|
1003
|
|
1004 if (bd_idx < 0) {
|
|
1005 av_log(ctx, AV_LOG_ERROR,
|
|
1006 "Invalid frame type VLC code, skipping\n");
|
|
1007 return -1;
|
|
1008 }
|
|
1009
|
|
1010 /* Pitch calculation for ACB_TYPE_ASYMMETRIC ("pitch-per-frame") */
|
|
1011 if (frame_descs[bd_idx].acb_type == ACB_TYPE_ASYMMETRIC) {
|
|
1012 /* Pitch is provided per frame, which is interpreted as the pitch of
|
|
1013 * the last sample of the last block of this frame. We can interpolate
|
|
1014 * the pitch of other blocks (and even pitch-per-sample) by gradually
|
|
1015 * incrementing/decrementing prev_frame_pitch to cur_pitch_val. */
|
|
1016 n_blocks_x2 = frame_descs[bd_idx].n_blocks << 1;
|
|
1017 log_n_blocks_x2 = frame_descs[bd_idx].log_n_blocks + 1;
|
|
1018 cur_pitch_val = s->min_pitch_val + get_bits(gb, s->pitch_nbits);
|
|
1019 cur_pitch_val = FFMIN(cur_pitch_val, s->max_pitch_val - 1);
|
|
1020 if (s->last_acb_type == ACB_TYPE_NONE ||
|
|
1021 20 * abs(cur_pitch_val - s->last_pitch_val) >
|
|
1022 (cur_pitch_val + s->last_pitch_val))
|
|
1023 s->last_pitch_val = cur_pitch_val;
|
|
1024
|
|
1025 /* pitch per block */
|
|
1026 for (n = 0; n < frame_descs[bd_idx].n_blocks; n++) {
|
|
1027 int fac = n * 2 + 1;
|
|
1028
|
|
1029 pitch[n] = (MUL16(fac, cur_pitch_val) +
|
|
1030 MUL16((n_blocks_x2 - fac), s->last_pitch_val) +
|
|
1031 frame_descs[bd_idx].n_blocks) >> log_n_blocks_x2;
|
|
1032 }
|
|
1033
|
|
1034 /* "pitch-diff-per-sample" for calculation of pitch per sample */
|
|
1035 s->pitch_diff_sh16 =
|
|
1036 ((cur_pitch_val - s->last_pitch_val) << 16) / MAX_FRAMESIZE;
|
|
1037 }
|
|
1038
|
|
1039 /* Global gain (if silence) and pitch-adaptive window coordinates */
|
|
1040 switch (frame_descs[bd_idx].fcb_type) {
|
|
1041 case FCB_TYPE_SILENCE:
|
|
1042 s->silence_gain = wmavoice_gain_silence[get_bits(gb, 8)];
|
|
1043 break;
|
|
1044 case FCB_TYPE_AW_PULSES:
|
|
1045 aw_parse_coords(s, gb, pitch);
|
|
1046 break;
|
|
1047 }
|
|
1048
|
|
1049 for (n = 0; n < frame_descs[bd_idx].n_blocks; n++) {
|
|
1050 int bl_pitch_sh2;
|
|
1051
|
|
1052 /* Pitch calculation for ACB_TYPE_HAMMING ("pitch-per-block") */
|
|
1053 switch (frame_descs[bd_idx].acb_type) {
|
|
1054 case ACB_TYPE_HAMMING: {
|
|
1055 /* Pitch is given per block. Per-block pitches are encoded as an
|
|
1056 * absolute value for the first block, and then delta values
|
|
1057 * relative to this value) for all subsequent blocks. The scale of
|
|
1058 * this pitch value is semi-logaritmic compared to its use in the
|
|
1059 * decoder, so we convert it to normal scale also. */
|
|
1060 int block_pitch,
|
|
1061 t1 = (s->block_conv_table[1] - s->block_conv_table[0]) << 2,
|
|
1062 t2 = (s->block_conv_table[2] - s->block_conv_table[1]) << 1,
|
|
1063 t3 = s->block_conv_table[3] - s->block_conv_table[2] + 1;
|
|
1064
|
|
1065 if (n == 0) {
|
|
1066 block_pitch = get_bits(gb, s->block_pitch_nbits);
|
|
1067 } else
|
|
1068 block_pitch = last_block_pitch - s->block_delta_pitch_hrange +
|
|
1069 get_bits(gb, s->block_delta_pitch_nbits);
|
|
1070 /* Convert last_ so that any next delta is within _range */
|
|
1071 last_block_pitch = av_clip(block_pitch,
|
|
1072 s->block_delta_pitch_hrange,
|
|
1073 s->block_pitch_range -
|
|
1074 s->block_delta_pitch_hrange);
|
|
1075
|
|
1076 /* Convert semi-log-style scale back to normal scale */
|
|
1077 if (block_pitch < t1) {
|
|
1078 bl_pitch_sh2 = (s->block_conv_table[0] << 2) + block_pitch;
|
|
1079 } else {
|
|
1080 block_pitch -= t1;
|
|
1081 if (block_pitch < t2) {
|
|
1082 bl_pitch_sh2 =
|
|
1083 (s->block_conv_table[1] << 2) + (block_pitch << 1);
|
|
1084 } else {
|
|
1085 block_pitch -= t2;
|
|
1086 if (block_pitch < t3) {
|
|
1087 bl_pitch_sh2 =
|
|
1088 (s->block_conv_table[2] + block_pitch) << 2;
|
|
1089 } else
|
|
1090 bl_pitch_sh2 = s->block_conv_table[3] << 2;
|
|
1091 }
|
|
1092 }
|
|
1093 pitch[n] = bl_pitch_sh2 >> 2;
|
|
1094 break;
|
|
1095 }
|
|
1096
|
|
1097 case ACB_TYPE_ASYMMETRIC: {
|
|
1098 bl_pitch_sh2 = pitch[n] << 2;
|
|
1099 break;
|
|
1100 }
|
|
1101
|
|
1102 default: // ACB_TYPE_NONE has no pitch
|
|
1103 bl_pitch_sh2 = 0;
|
|
1104 break;
|
|
1105 }
|
|
1106
|
|
1107 synth_block(s, gb, n, block_nsamples, bl_pitch_sh2,
|
|
1108 lsps, prev_lsps, &frame_descs[bd_idx],
|
|
1109 &excitation[n * block_nsamples],
|
|
1110 &synth[n * block_nsamples]);
|
|
1111 }
|
|
1112
|
|
1113 /* Averaging projection filter, if applicable. Else, just copy samples
|
|
1114 * from synthesis buffer */
|
|
1115 if (s->do_apf) {
|
|
1116 // FIXME this is where APF would take place, currently not implemented
|
|
1117 av_log_missing_feature(ctx, "APF", 0);
|
|
1118 s->do_apf = 0;
|
|
1119 } //else
|
|
1120 for (n = 0; n < 160; n++)
|
|
1121 samples[n] = av_clipf(synth[n], -1.0, 1.0);
|
|
1122
|
|
1123 /* Cache values for next frame */
|
|
1124 s->frame_cntr++;
|
|
1125 if (s->frame_cntr >= 0xFFFF) s->frame_cntr -= 0xFFFF; // i.e. modulo (%)
|
|
1126 s->last_acb_type = frame_descs[bd_idx].acb_type;
|
|
1127 switch (frame_descs[bd_idx].acb_type) {
|
|
1128 case ACB_TYPE_NONE:
|
|
1129 s->last_pitch_val = 0;
|
|
1130 break;
|
|
1131 case ACB_TYPE_ASYMMETRIC:
|
|
1132 s->last_pitch_val = cur_pitch_val;
|
|
1133 break;
|
|
1134 case ACB_TYPE_HAMMING:
|
|
1135 s->last_pitch_val = pitch[frame_descs[bd_idx].n_blocks - 1];
|
|
1136 break;
|
|
1137 }
|
|
1138
|
|
1139 return 0;
|
|
1140 }
|
|
1141
|
|
1142 /**
|
|
1143 * Ensure minimum value for first item, maximum value for last value,
|
|
1144 * proper spacing between each value and proper ordering.
|
|
1145 *
|
|
1146 * @param lsps array of LSPs
|
|
1147 * @param num size of LSP array
|
|
1148 *
|
|
1149 * @note basically a double version of #ff_acelp_reorder_lsf(), might be
|
|
1150 * useful to put in a generic location later on. Parts are also
|
|
1151 * present in #ff_set_min_dist_lsf() + #ff_sort_nearly_sorted_floats(),
|
|
1152 * which is in float.
|
|
1153 */
|
|
1154 static void stabilize_lsps(double *lsps, int num)
|
|
1155 {
|
|
1156 int n, m, l;
|
|
1157
|
|
1158 /* set minimum value for first, maximum value for last and minimum
|
|
1159 * spacing between LSF values.
|
|
1160 * Very similar to ff_set_min_dist_lsf(), but in double. */
|
|
1161 lsps[0] = FFMAX(lsps[0], 0.0015 * M_PI);
|
|
1162 for (n = 1; n < num; n++)
|
|
1163 lsps[n] = FFMAX(lsps[n], lsps[n - 1] + 0.0125 * M_PI);
|
|
1164 lsps[num - 1] = FFMIN(lsps[num - 1], 0.9985 * M_PI);
|
|
1165
|
|
1166 /* reorder (looks like one-time / non-recursed bubblesort).
|
|
1167 * Very similar to ff_sort_nearly_sorted_floats(), but in double. */
|
|
1168 for (n = 1; n < num; n++) {
|
|
1169 if (lsps[n] < lsps[n - 1]) {
|
|
1170 for (m = 1; m < num; m++) {
|
|
1171 double tmp = lsps[m];
|
|
1172 for (l = m - 1; l >= 0; l--) {
|
|
1173 if (lsps[l] <= tmp) break;
|
|
1174 lsps[l + 1] = lsps[l];
|
|
1175 }
|
|
1176 lsps[l + 1] = tmp;
|
|
1177 }
|
|
1178 break;
|
|
1179 }
|
|
1180 }
|
|
1181 }
|
|
1182
|
|
1183 /**
|
|
1184 * Test if there's enough bits to read 1 superframe.
|
|
1185 *
|
|
1186 * @param orig_gb bit I/O context used for reading. This function
|
|
1187 * does not modify the state of the bitreader; it
|
|
1188 * only uses it to copy the current stream position
|
|
1189 * @param s WMA Voice decoding context private data
|
|
1190 * @returns -1 if unsupported, 1 on not enough bits or 0 if OK.
|
|
1191 */
|
|
1192 static int check_bits_for_superframe(GetBitContext *orig_gb,
|
|
1193 WMAVoiceContext *s)
|
|
1194 {
|
|
1195 GetBitContext s_gb, *gb = &s_gb;
|
|
1196 int n, need_bits, bd_idx;
|
|
1197 const struct frame_type_desc *frame_desc;
|
|
1198
|
|
1199 /* initialize a copy */
|
|
1200 init_get_bits(gb, orig_gb->buffer, orig_gb->size_in_bits);
|
|
1201 skip_bits_long(gb, get_bits_count(orig_gb));
|
|
1202 assert(get_bits_left(gb) == get_bits_left(orig_gb));
|
|
1203
|
|
1204 /* superframe header */
|
|
1205 if (get_bits_left(gb) < 14)
|
|
1206 return 1;
|
|
1207 if (!get_bits1(gb))
|
|
1208 return -1; // WMAPro-in-WMAVoice superframe
|
|
1209 if (get_bits1(gb)) skip_bits(gb, 12); // number of samples in superframe
|
|
1210 if (s->has_residual_lsps) { // residual LSPs (for all frames)
|
|
1211 if (get_bits_left(gb) < s->sframe_lsp_bitsize)
|
|
1212 return 1;
|
|
1213 skip_bits_long(gb, s->sframe_lsp_bitsize);
|
|
1214 }
|
|
1215
|
|
1216 /* frames */
|
|
1217 for (n = 0; n < MAX_FRAMES; n++) {
|
|
1218 int aw_idx_is_ext = 0;
|
|
1219
|
|
1220 if (!s->has_residual_lsps) { // independent LSPs (per-frame)
|
|
1221 if (get_bits_left(gb) < s->frame_lsp_bitsize) return 1;
|
|
1222 skip_bits_long(gb, s->frame_lsp_bitsize);
|
|
1223 }
|
|
1224 bd_idx = s->vbm_tree[get_vlc2(gb, frame_type_vlc.table, 6, 3)];
|
|
1225 if (bd_idx < 0)
|
|
1226 return -1; // invalid frame type VLC code
|
|
1227 frame_desc = &frame_descs[bd_idx];
|
|
1228 if (frame_desc->acb_type == ACB_TYPE_ASYMMETRIC) {
|
|
1229 if (get_bits_left(gb) < s->pitch_nbits)
|
|
1230 return 1;
|
|
1231 skip_bits_long(gb, s->pitch_nbits);
|
|
1232 }
|
|
1233 if (frame_desc->fcb_type == FCB_TYPE_SILENCE) {
|
|
1234 skip_bits(gb, 8);
|
|
1235 } else if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) {
|
|
1236 int tmp = get_bits(gb, 6);
|
|
1237 if (tmp >= 0x36) {
|
|
1238 skip_bits(gb, 2);
|
|
1239 aw_idx_is_ext = 1;
|
|
1240 }
|
|
1241 }
|
|
1242
|
|
1243 /* blocks */
|
|
1244 if (frame_desc->acb_type == ACB_TYPE_HAMMING) {
|
|
1245 need_bits = s->block_pitch_nbits +
|
|
1246 (frame_desc->n_blocks - 1) * s->block_delta_pitch_nbits;
|
|
1247 } else if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) {
|
|
1248 need_bits = 2 * !aw_idx_is_ext;
|
|
1249 } else
|
|
1250 need_bits = 0;
|
|
1251 need_bits += frame_desc->frame_size;
|
|
1252 if (get_bits_left(gb) < need_bits)
|
|
1253 return 1;
|
|
1254 skip_bits_long(gb, need_bits);
|
|
1255 }
|
|
1256
|
|
1257 return 0;
|
|
1258 }
|
|
1259
|
|
1260 /**
|
|
1261 * Synthesize output samples for a single superframe. If we have any data
|
|
1262 * cached in s->sframe_cache, that will be used instead of whatever is loaded
|
|
1263 * in s->gb.
|
|
1264 *
|
|
1265 * WMA Voice superframes contain 3 frames, each containing 160 audio samples,
|
|
1266 * to give a total of 480 samples per frame. See #synth_frame() for frame
|
|
1267 * parsing. In addition to 3 frames, superframes can also contain the LSPs
|
|
1268 * (if these are globally specified for all frames (residually); they can
|
|
1269 * also be specified individually per-frame. See the s->has_residual_lsps
|
|
1270 * option), and can specify the number of samples encoded in this superframe
|
|
1271 * (if less than 480), usually used to prevent blanks at track boundaries.
|
|
1272 *
|
|
1273 * @param ctx WMA Voice decoder context
|
|
1274 * @param samples pointer to output buffer for voice samples
|
|
1275 * @param data_size pointer containing the size of #samples on input, and the
|
|
1276 * amount of #samples filled on output
|
|
1277 * @return 0 on success, <0 on error or 1 if there was not enough data to
|
|
1278 * fully parse the superframe
|
|
1279 */
|
|
1280 static int synth_superframe(AVCodecContext *ctx,
|
|
1281 float *samples, int *data_size)
|
|
1282 {
|
|
1283 WMAVoiceContext *s = ctx->priv_data;
|
|
1284 GetBitContext *gb = &s->gb, s_gb;
|
|
1285 int n, res, n_samples = 480;
|
|
1286 double lsps[MAX_FRAMES][MAX_LSPS];
|
|
1287 const double *mean_lsf = s->lsps == 16 ?
|
|
1288 wmavoice_mean_lsf16[s->lsp_def_mode] : wmavoice_mean_lsf10[s->lsp_def_mode];
|
|
1289 float excitation[MAX_SIGNAL_HISTORY + MAX_SFRAMESIZE + 12];
|
|
1290 float synth[MAX_LSPS + MAX_SFRAMESIZE];
|
|
1291
|
|
1292 memcpy(synth, s->synth_history,
|
|
1293 s->lsps * sizeof(*synth));
|
|
1294 memcpy(excitation, s->excitation_history,
|
|
1295 s->history_nsamples * sizeof(*excitation));
|
|
1296
|
|
1297 if (s->sframe_cache_size > 0) {
|
|
1298 gb = &s_gb;
|
|
1299 init_get_bits(gb, s->sframe_cache, s->sframe_cache_size);
|
|
1300 s->sframe_cache_size = 0;
|
|
1301 }
|
|
1302
|
|
1303 if ((res = check_bits_for_superframe(gb, s)) == 1) return 1;
|
|
1304
|
|
1305 /* First bit is speech/music bit, it differentiates between WMAVoice
|
|
1306 * speech samples (the actual codec) and WMAVoice music samples, which
|
|
1307 * are really WMAPro-in-WMAVoice-superframes. I've never seen those in
|
|
1308 * the wild yet. */
|
|
1309 if (!get_bits1(gb)) {
|
|
1310 av_log_missing_feature(ctx, "WMAPro-in-WMAVoice support", 1);
|
|
1311 return -1;
|
|
1312 }
|
|
1313
|
|
1314 /* (optional) nr. of samples in superframe; always <= 480 and >= 0 */
|
|
1315 if (get_bits1(gb)) {
|
|
1316 if ((n_samples = get_bits(gb, 12)) > 480) {
|
|
1317 av_log(ctx, AV_LOG_ERROR,
|
|
1318 "Superframe encodes >480 samples (%d), not allowed\n",
|
|
1319 n_samples);
|
|
1320 return -1;
|
|
1321 }
|
|
1322 }
|
|
1323 /* Parse LSPs, if global for the superframe (can also be per-frame). */
|
|
1324 if (s->has_residual_lsps) {
|
|
1325 double prev_lsps[MAX_LSPS], a1[MAX_LSPS * 2], a2[MAX_LSPS * 2];
|
|
1326
|
|
1327 for (n = 0; n < s->lsps; n++)
|
|
1328 prev_lsps[n] = s->prev_lsps[n] - mean_lsf[n];
|
|
1329
|
|
1330 if (s->lsps == 10) {
|
|
1331 dequant_lsp10r(gb, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode);
|
|
1332 } else /* s->lsps == 16 */
|
|
1333 dequant_lsp16r(gb, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode);
|
|
1334
|
|
1335 for (n = 0; n < s->lsps; n++) {
|
|
1336 lsps[0][n] = mean_lsf[n] + (a1[n] - a2[n * 2]);
|
|
1337 lsps[1][n] = mean_lsf[n] + (a1[s->lsps + n] - a2[n * 2 + 1]);
|
|
1338 lsps[2][n] += mean_lsf[n];
|
|
1339 }
|
|
1340 for (n = 0; n < 3; n++)
|
|
1341 stabilize_lsps(lsps[n], s->lsps);
|
|
1342 }
|
|
1343
|
|
1344 /* Parse frames, optionally preceeded by per-frame (independent) LSPs. */
|
|
1345 for (n = 0; n < 3; n++) {
|
|
1346 if (!s->has_residual_lsps) {
|
|
1347 int m;
|
|
1348
|
|
1349 if (s->lsps == 10) {
|
|
1350 dequant_lsp10i(gb, lsps[n]);
|
|
1351 } else /* s->lsps == 16 */
|
|
1352 dequant_lsp16i(gb, lsps[n]);
|
|
1353
|
|
1354 for (m = 0; m < s->lsps; m++)
|
|
1355 lsps[n][m] += mean_lsf[m];
|
|
1356 stabilize_lsps(lsps[n], s->lsps);
|
|
1357 }
|
|
1358
|
|
1359 if ((res = synth_frame(ctx, gb,
|
|
1360 &samples[n * MAX_FRAMESIZE],
|
|
1361 lsps[n], n == 0 ? s->prev_lsps : lsps[n - 1],
|
|
1362 &excitation[s->history_nsamples + n * MAX_FRAMESIZE],
|
|
1363 &synth[s->lsps + n * MAX_FRAMESIZE])))
|
|
1364 return res;
|
|
1365 }
|
|
1366
|
|
1367 /* Statistics? FIXME - we don't check for length, a slight overrun
|
|
1368 * will be caught by internal buffer padding, and anything else
|
|
1369 * will be skipped, not read. */
|
|
1370 if (get_bits1(gb)) {
|
|
1371 res = get_bits(gb, 4);
|
|
1372 skip_bits(gb, 10 * (res + 1));
|
|
1373 }
|
|
1374
|
|
1375 /* Specify nr. of output samples */
|
|
1376 *data_size = n_samples * sizeof(float);
|
|
1377
|
|
1378 /* Update history */
|
|
1379 memcpy(s->prev_lsps, lsps[2],
|
|
1380 s->lsps * sizeof(*s->prev_lsps));
|
|
1381 memcpy(s->synth_history, &synth[MAX_SFRAMESIZE],
|
|
1382 s->lsps * sizeof(*synth));
|
|
1383 memcpy(s->excitation_history, &excitation[MAX_SFRAMESIZE],
|
|
1384 s->history_nsamples * sizeof(*excitation));
|
|
1385
|
|
1386 return 0;
|
|
1387 }
|
|
1388
|
|
1389 /**
|
|
1390 * Parse the packet header at the start of each packet (input data to this
|
|
1391 * decoder).
|
|
1392 *
|
|
1393 * @param s WMA Voice decoding context private data
|
|
1394 * @returns 1 if not enough bits were available, or 0 on success.
|
|
1395 */
|
|
1396 static int parse_packet_header(WMAVoiceContext *s)
|
|
1397 {
|
|
1398 GetBitContext *gb = &s->gb;
|
|
1399 unsigned int res;
|
|
1400
|
|
1401 if (get_bits_left(gb) < 11)
|
|
1402 return 1;
|
|
1403 skip_bits(gb, 4); // packet sequence number
|
|
1404 s->has_residual_lsps = get_bits1(gb);
|
|
1405 do {
|
|
1406 res = get_bits(gb, 6); // number of superframes per packet
|
|
1407 // (minus first one if there is spillover)
|
|
1408 if (get_bits_left(gb) < 6 * (res == 0x3F) + s->spillover_bitsize)
|
|
1409 return 1;
|
|
1410 } while (res == 0x3F);
|
|
1411 s->spillover_nbits = get_bits(gb, s->spillover_bitsize);
|
|
1412
|
|
1413 return 0;
|
|
1414 }
|
|
1415
|
|
1416 /**
|
|
1417 * Copy (unaligned) bits from gb/data/size to pb.
|
|
1418 *
|
|
1419 * @param pb target buffer to copy bits into
|
|
1420 * @param data source buffer to copy bits from
|
|
1421 * @param size size of the source data, in bytes
|
|
1422 * @param gb bit I/O context specifying the current position in the source.
|
|
1423 * data. This function might use this to align the bit position to
|
|
1424 * a whole-byte boundary before calling #ff_copy_bits() on aligned
|
|
1425 * source data
|
|
1426 * @param nbits the amount of bits to copy from source to target
|
|
1427 *
|
|
1428 * @note after calling this function, the current position in the input bit
|
|
1429 * I/O context is undefined.
|
|
1430 */
|
|
1431 static void copy_bits(PutBitContext *pb,
|
|
1432 const uint8_t *data, int size,
|
|
1433 GetBitContext *gb, int nbits)
|
|
1434 {
|
|
1435 int rmn_bytes, rmn_bits;
|
|
1436
|
|
1437 rmn_bits = rmn_bytes = get_bits_left(gb);
|
|
1438 if (rmn_bits < nbits)
|
|
1439 return;
|
|
1440 rmn_bits &= 7; rmn_bytes >>= 3;
|
|
1441 if ((rmn_bits = FFMIN(rmn_bits, nbits)) > 0)
|
|
1442 put_bits(pb, rmn_bits, get_bits(gb, rmn_bits));
|
|
1443 ff_copy_bits(pb, data + size - rmn_bytes,
|
|
1444 FFMIN(nbits - rmn_bits, rmn_bytes << 3));
|
|
1445 }
|
|
1446
|
|
1447 /**
|
|
1448 * Packet decoding: a packet is anything that the (ASF) demuxer contains,
|
|
1449 * and we expect that the demuxer / application provides it to us as such
|
|
1450 * (else you'll probably get garbage as output). Every packet has a size of
|
|
1451 * ctx->block_align bytes, starts with a packet header (see
|
|
1452 * #parse_packet_header()), and then a series of superframes. Superframe
|
|
1453 * boundaries may exceed packets, i.e. superframes can split data over
|
|
1454 * multiple (two) packets.
|
|
1455 *
|
|
1456 * For more information about frames, see #synth_superframe().
|
|
1457 */
|
|
1458 static int wmavoice_decode_packet(AVCodecContext *ctx, void *data,
|
|
1459 int *data_size, AVPacket *avpkt)
|
|
1460 {
|
|
1461 WMAVoiceContext *s = ctx->priv_data;
|
|
1462 GetBitContext *gb = &s->gb;
|
|
1463 int size, res, pos;
|
|
1464
|
|
1465 if (*data_size < 480 * sizeof(float)) {
|
|
1466 av_log(ctx, AV_LOG_ERROR,
|
|
1467 "Output buffer too small (%d given - %lu needed)\n",
|
|
1468 *data_size, 480 * sizeof(float));
|
|
1469 return -1;
|
|
1470 }
|
|
1471 *data_size = 0;
|
|
1472
|
|
1473 /* Packets are sometimes a multiple of ctx->block_align, with a packet
|
|
1474 * header at each ctx->block_align bytes. However, FFmpeg's ASF demuxer
|
|
1475 * feeds us ASF packets, which may concatenate multiple "codec" packets
|
|
1476 * in a single "muxer" packet, so we artificially emulate that by
|
|
1477 * capping the packet size at ctx->block_align. */
|
|
1478 for (size = avpkt->size; size > ctx->block_align; size -= ctx->block_align);
|
|
1479 if (!size)
|
|
1480 return 0;
|
|
1481 init_get_bits(&s->gb, avpkt->data, size << 3);
|
|
1482
|
|
1483 /* size == ctx->block_align is used to indicate whether we are dealing with
|
|
1484 * a new packet or a packet of which we already read the packet header
|
|
1485 * previously. */
|
|
1486 if (size == ctx->block_align) { // new packet header
|
|
1487 if ((res = parse_packet_header(s)) < 0)
|
|
1488 return res;
|
|
1489
|
|
1490 /* If the packet header specifies a s->spillover_nbits, then we want
|
|
1491 * to push out all data of the previous packet (+ spillover) before
|
|
1492 * continuing to parse new superframes in the current packet. */
|
|
1493 if (s->spillover_nbits > 0) {
|
|
1494 if (s->sframe_cache_size > 0) {
|
|
1495 int cnt = get_bits_count(gb);
|
|
1496 copy_bits(&s->pb, avpkt->data, size, gb, s->spillover_nbits);
|
|
1497 flush_put_bits(&s->pb);
|
|
1498 s->sframe_cache_size += s->spillover_nbits;
|
|
1499 if ((res = synth_superframe(ctx, data, data_size)) == 0 &&
|
|
1500 *data_size > 0) {
|
|
1501 cnt += s->spillover_nbits;
|
|
1502 s->skip_bits_next = cnt & 7;
|
|
1503 return cnt >> 3;
|
|
1504 } else
|
|
1505 skip_bits_long (gb, s->spillover_nbits - cnt +
|
|
1506 get_bits_count(gb)); // resync
|
|
1507 } else
|
|
1508 skip_bits_long(gb, s->spillover_nbits); // resync
|
|
1509 }
|
|
1510 } else if (s->skip_bits_next)
|
|
1511 skip_bits(gb, s->skip_bits_next);
|
|
1512
|
|
1513 /* Try parsing superframes in current packet */
|
|
1514 s->sframe_cache_size = 0;
|
|
1515 s->skip_bits_next = 0;
|
|
1516 pos = get_bits_left(gb);
|
|
1517 if ((res = synth_superframe(ctx, data, data_size)) < 0) {
|
|
1518 return res;
|
|
1519 } else if (*data_size > 0) {
|
|
1520 int cnt = get_bits_count(gb);
|
|
1521 s->skip_bits_next = cnt & 7;
|
|
1522 return cnt >> 3;
|
|
1523 } else if ((s->sframe_cache_size = pos) > 0) {
|
|
1524 /* rewind bit reader to start of last (incomplete) superframe... */
|
|
1525 init_get_bits(gb, avpkt->data, size << 3);
|
|
1526 skip_bits_long(gb, (size << 3) - pos);
|
|
1527 assert(get_bits_left(gb) == pos);
|
|
1528
|
|
1529 /* ...and cache it for spillover in next packet */
|
|
1530 init_put_bits(&s->pb, s->sframe_cache, SFRAME_CACHE_MAXSIZE);
|
|
1531 copy_bits(&s->pb, avpkt->data, size, gb, s->sframe_cache_size);
|
|
1532 // FIXME bad - just copy bytes as whole and add use the
|
|
1533 // skip_bits_next field
|
|
1534 }
|
|
1535
|
|
1536 return size;
|
|
1537 }
|
|
1538
|
|
1539 static av_cold void wmavoice_flush(AVCodecContext *ctx)
|
|
1540 {
|
|
1541 WMAVoiceContext *s = ctx->priv_data;
|
|
1542 int n;
|
|
1543
|
|
1544 s->sframe_cache_size = 0;
|
|
1545 s->skip_bits_next = 0;
|
|
1546 for (n = 0; n < s->lsps; n++)
|
|
1547 s->prev_lsps[n] = M_PI * (n + 1.0) / (s->lsps + 1.0);
|
|
1548 memset(s->excitation_history, 0,
|
|
1549 sizeof(*s->excitation_history) * MAX_SIGNAL_HISTORY);
|
|
1550 memset(s->synth_history, 0,
|
|
1551 sizeof(*s->synth_history) * MAX_LSPS);
|
|
1552 memset(s->gain_pred_err, 0,
|
|
1553 sizeof(s->gain_pred_err));
|
|
1554 }
|
|
1555
|
|
1556 AVCodec wmavoice_decoder = {
|
|
1557 "wmavoice",
|
|
1558 CODEC_TYPE_AUDIO,
|
|
1559 CODEC_ID_WMAVOICE,
|
|
1560 sizeof(WMAVoiceContext),
|
|
1561 wmavoice_decode_init,
|
|
1562 NULL,
|
|
1563 NULL,
|
|
1564 wmavoice_decode_packet,
|
|
1565 CODEC_CAP_SUBFRAMES,
|
|
1566 .flush = wmavoice_flush,
|
|
1567 .long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio Voice"),
|
|
1568 };
|