2914
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1 /*
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2 * QDM2 compatible decoder
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3 * Copyright (c) 2003 Ewald Snel
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4 * Copyright (c) 2005 Benjamin Larsson
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5 * Copyright (c) 2005 Alex Beregszaszi
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6 * Copyright (c) 2005 Roberto Togni
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7 *
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8 * This library 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 of the License, or (at your option) any later version.
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12 *
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13 * This library 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 this library; if not, write to the Free Software
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20 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 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 qdm2.c
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26 * QDM2 decoder
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27 * @author Ewald Snel, Benjamin Larsson, Alex Beregszaszi, Roberto Togni
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28 * The decoder is not perfect yet, there are still some distorions expecially
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29 * on files encoded with 16 or 8 subbands
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30 */
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31
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32 #include <math.h>
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33 #include <stddef.h>
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34 #include <stdio.h>
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35
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36 #define ALT_BITSTREAM_READER_LE
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37 #include "avcodec.h"
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38 #include "bitstream.h"
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39 #include "dsputil.h"
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40
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41 #ifdef CONFIG_MPEGAUDIO_HP
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42 #define USE_HIGHPRECISION
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43 #endif
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44
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45 #include "mpegaudio.h"
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46
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47 #include "qdm2data.h"
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48
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49 #undef NDEBUG
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50 #include <assert.h>
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51
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52
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53 #define SOFTCLIP_THRESHOLD 27600
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54 #define HARDCLIP_THRESHOLD 35716
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55
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56
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57 #define QDM2_LIST_ADD(list, size, packet) \
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58 do { \
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59 if (size > 0) { \
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60 list[size - 1].next = &list[size]; \
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61 } \
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62 list[size].packet = packet; \
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63 list[size].next = NULL; \
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64 size++; \
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65 } while(0)
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66
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67 // Result is 8, 16 or 30
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68 #define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling))
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69
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70 #define FIX_NOISE_IDX(noise_idx) \
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71 if ((noise_idx) >= 3840) \
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72 (noise_idx) -= 3840; \
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73
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74 #define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)])
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75
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76 #define BITS_LEFT(length,gb) ((length) - get_bits_count ((gb)))
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77
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78 #define SAMPLES_NEEDED \
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79 av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n");
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80
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81 #define SAMPLES_NEEDED_2(why) \
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82 av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why);
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83
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84
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85 typedef int8_t sb_int8_array[2][30][64];
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86
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87 /**
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88 * Subpacket
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89 */
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90 typedef struct {
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91 int type; ///< subpacket type
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92 unsigned int size; ///< subpacket size
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93 const uint8_t *data; ///< pointer to subpacket data (points to input data buffer, it's not a private copy)
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94 } QDM2SubPacket;
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95
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96 /**
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97 * A node in subpacket list
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98 */
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99 typedef struct _QDM2SubPNode {
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100 QDM2SubPacket *packet; ///< packet
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101 struct _QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node
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102 } QDM2SubPNode;
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103
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104 typedef struct {
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105 float level;
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106 float *samples_im;
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107 float *samples_re;
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108 float *table;
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109 int phase;
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110 int phase_shift;
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111 int duration;
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112 short time_index;
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113 short cutoff;
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114 } FFTTone;
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115
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116 typedef struct {
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117 int16_t sub_packet;
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118 uint8_t channel;
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119 int16_t offset;
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120 int16_t exp;
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121 uint8_t phase;
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122 } FFTCoefficient;
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123
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124 typedef struct {
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125 float re;
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126 float im;
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127 } QDM2Complex;
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128
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129 typedef struct {
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130 QDM2Complex complex[256 + 1] __attribute__((aligned(16)));
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131 float samples_im[MPA_MAX_CHANNELS][256];
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132 float samples_re[MPA_MAX_CHANNELS][256];
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133 } QDM2FFT;
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134
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135 /**
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136 * QDM2 decoder context
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137 */
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138 typedef struct {
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139 /// Parameters from codec header, do not change during playback
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140 int nb_channels; ///< number of channels
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141 int channels; ///< number of channels
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142 int group_size; ///< size of frame group (16 frames per group)
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143 int fft_size; ///< size of FFT, in complex numbers
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144 int checksum_size; ///< size of data block, used also for checksum
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145
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146 /// Parameters built from header parameters, do not change during playback
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147 int group_order; ///< order of frame group
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148 int fft_order; ///< order of FFT (actually fftorder+1)
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149 int fft_frame_size; ///< size of fft frame, in components (1 comples = re + im)
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150 int frame_size; ///< size of data frame
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151 int frequency_range;
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152 int sub_sampling; ///< subsampling: 0=25%, 1=50%, 2=100% */
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153 int coeff_per_sb_select; ///< selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2
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154 int cm_table_select; ///< selector for "coding method" tables. Can be 0, 1 (from init: 0-4)
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155
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156 /// Packets and packet lists
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157 QDM2SubPacket sub_packets[16]; ///< the packets themselves
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158 QDM2SubPNode sub_packet_list_A[16]; ///< list of all packets
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159 QDM2SubPNode sub_packet_list_B[16]; ///< FFT packets B are on list
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160 int sub_packets_B; ///< number of packets on 'B' list
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161 QDM2SubPNode sub_packet_list_C[16]; ///< packets with errors?
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162 QDM2SubPNode sub_packet_list_D[16]; ///< DCT packets
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163
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164 /// FFT and tones
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165 FFTTone fft_tones[1000];
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166 int fft_tone_start;
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167 int fft_tone_end;
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168 FFTCoefficient fft_coefs[1000];
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169 int fft_coefs_index;
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170 int fft_coefs_min_index[5];
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171 int fft_coefs_max_index[5];
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172 int fft_level_exp[6];
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173 FFTContext fft_ctx;
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174 FFTComplex exptab[128];
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175 QDM2FFT fft;
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176
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177 /// I/O data
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178 uint8_t *compressed_data;
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179 int compressed_size;
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180 float output_buffer[1024];
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181
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182 /// Synthesis filter
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183 MPA_INT synth_buf[MPA_MAX_CHANNELS][512*2] __attribute__((aligned(16)));
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184 int synth_buf_offset[MPA_MAX_CHANNELS];
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185 int32_t sb_samples[MPA_MAX_CHANNELS][128][SBLIMIT] __attribute__((aligned(16)));
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186
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187 /// Mixed temporary data used in decoding
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188 float tone_level[MPA_MAX_CHANNELS][30][64];
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189 int8_t coding_method[MPA_MAX_CHANNELS][30][64];
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190 int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8];
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191 int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8];
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192 int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8];
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193 int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8];
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194 int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26];
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195 int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64];
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196 int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64];
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197
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198 // Flags
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199 int has_errors; ///< packet have errors
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200 int superblocktype_2_3; ///< select fft tables and some algorithm based on superblock type
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201 int do_synth_filter; ///< used to perform or skip synthesis filter
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202
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203 int sub_packet;
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204 int noise_idx; ///< Index for dithering noise table
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205 } QDM2Context;
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206
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207
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208 static uint8_t empty_buffer[FF_INPUT_BUFFER_PADDING_SIZE];
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209
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210 static VLC vlc_tab_level;
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211 static VLC vlc_tab_diff;
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212 static VLC vlc_tab_run;
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213 static VLC fft_level_exp_alt_vlc;
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214 static VLC fft_level_exp_vlc;
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215 static VLC fft_stereo_exp_vlc;
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216 static VLC fft_stereo_phase_vlc;
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217 static VLC vlc_tab_tone_level_idx_hi1;
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218 static VLC vlc_tab_tone_level_idx_mid;
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219 static VLC vlc_tab_tone_level_idx_hi2;
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220 static VLC vlc_tab_type30;
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221 static VLC vlc_tab_type34;
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222 static VLC vlc_tab_fft_tone_offset[5];
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223
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224 static uint16_t softclip_table[HARDCLIP_THRESHOLD - SOFTCLIP_THRESHOLD + 1];
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225 static float noise_table[4096];
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226 static uint8_t random_dequant_index[256][5];
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227 static uint8_t random_dequant_type24[128][3];
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228 static float noise_samples[128];
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229
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230 static MPA_INT mpa_window[512] __attribute__((aligned(16)));
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231
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232
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233 static void softclip_table_init() {
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234 int i;
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235 double dfl = SOFTCLIP_THRESHOLD - 32767;
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236 float delta = 1.0 / -dfl;
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237 for (i = 0; i < HARDCLIP_THRESHOLD - SOFTCLIP_THRESHOLD + 1; i++)
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238 softclip_table[i] = SOFTCLIP_THRESHOLD - ((int)(sin((float)i * delta) * dfl) & 0x0000FFFF);
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239 }
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240
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241
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242 // random generated table
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243 static void rnd_table_init() {
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244 int i,j;
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245 uint32_t ldw,hdw;
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246 uint64_t tmp64_1;
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247 uint64_t random_seed = 0;
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248 float delta = 1.0 / 16384.0;
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249 for(i = 0; i < 4096 ;i++) {
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250 random_seed = random_seed * 214013 + 2531011;
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251 noise_table[i] = (delta * (float)(((int32_t)random_seed >> 16) & 0x00007FFF)- 1.0) * 1.3;
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252 }
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253
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254 for (i = 0; i < 256 ;i++) {
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255 random_seed = 81;
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256 ldw = i;
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257 for (j = 0; j < 5 ;j++) {
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258 random_dequant_index[i][j] = (uint8_t)((ldw / random_seed) & 0xFF);
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259 ldw = (uint32_t)ldw % (uint32_t)random_seed;
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260 tmp64_1 = (random_seed * 0x55555556);
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261 hdw = (uint32_t)(tmp64_1 >> 32);
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262 random_seed = (uint64_t)(hdw + (ldw >> 31));
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263 }
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264 }
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265 for (i = 0; i < 128 ;i++) {
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266 random_seed = 25;
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267 ldw = i;
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268 for (j = 0; j < 3 ;j++) {
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269 random_dequant_type24[i][j] = (uint8_t)((ldw / random_seed) & 0xFF);
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270 ldw = (uint32_t)ldw % (uint32_t)random_seed;
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271 tmp64_1 = (random_seed * 0x66666667);
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272 hdw = (uint32_t)(tmp64_1 >> 33);
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273 random_seed = hdw + (ldw >> 31);
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274 }
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275 }
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276 }
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277
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278
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279 static void init_noise_samples() {
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280 int i;
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281 int random_seed = 0;
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282 float delta = 1.0 / 16384.0;
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283 for (i = 0; i < 128;i++) {
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284 random_seed = random_seed * 214013 + 2531011;
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285 noise_samples[i] = (delta * (float)((random_seed >> 16) & 0x00007fff) - 1.0);
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286 }
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287 }
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288
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289
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290 static void qdm2_init_vlc()
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291 {
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292 init_vlc (&vlc_tab_level, 8, 24,
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293 vlc_tab_level_huffbits, 1, 1,
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294 vlc_tab_level_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
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295
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296 init_vlc (&vlc_tab_diff, 8, 37,
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297 vlc_tab_diff_huffbits, 1, 1,
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298 vlc_tab_diff_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
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299
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300 init_vlc (&vlc_tab_run, 5, 6,
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301 vlc_tab_run_huffbits, 1, 1,
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302 vlc_tab_run_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
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303
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304 init_vlc (&fft_level_exp_alt_vlc, 8, 28,
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305 fft_level_exp_alt_huffbits, 1, 1,
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306 fft_level_exp_alt_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
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307
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308 init_vlc (&fft_level_exp_vlc, 8, 20,
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309 fft_level_exp_huffbits, 1, 1,
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310 fft_level_exp_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
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311
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312 init_vlc (&fft_stereo_exp_vlc, 6, 7,
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313 fft_stereo_exp_huffbits, 1, 1,
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314 fft_stereo_exp_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
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315
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316 init_vlc (&fft_stereo_phase_vlc, 6, 9,
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317 fft_stereo_phase_huffbits, 1, 1,
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318 fft_stereo_phase_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
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319
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320 init_vlc (&vlc_tab_tone_level_idx_hi1, 8, 20,
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321 vlc_tab_tone_level_idx_hi1_huffbits, 1, 1,
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322 vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
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323
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324 init_vlc (&vlc_tab_tone_level_idx_mid, 8, 24,
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325 vlc_tab_tone_level_idx_mid_huffbits, 1, 1,
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326 vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
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327
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328 init_vlc (&vlc_tab_tone_level_idx_hi2, 8, 24,
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329 vlc_tab_tone_level_idx_hi2_huffbits, 1, 1,
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330 vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
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331
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332 init_vlc (&vlc_tab_type30, 6, 9,
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333 vlc_tab_type30_huffbits, 1, 1,
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334 vlc_tab_type30_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
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335
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336 init_vlc (&vlc_tab_type34, 5, 10,
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337 vlc_tab_type34_huffbits, 1, 1,
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338 vlc_tab_type34_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
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339
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340 init_vlc (&vlc_tab_fft_tone_offset[0], 8, 23,
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341 vlc_tab_fft_tone_offset_0_huffbits, 1, 1,
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342 vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
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343
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344 init_vlc (&vlc_tab_fft_tone_offset[1], 8, 28,
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345 vlc_tab_fft_tone_offset_1_huffbits, 1, 1,
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346 vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
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347
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348 init_vlc (&vlc_tab_fft_tone_offset[2], 8, 32,
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349 vlc_tab_fft_tone_offset_2_huffbits, 1, 1,
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350 vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
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351
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352 init_vlc (&vlc_tab_fft_tone_offset[3], 8, 35,
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353 vlc_tab_fft_tone_offset_3_huffbits, 1, 1,
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354 vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
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355
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356 init_vlc (&vlc_tab_fft_tone_offset[4], 8, 38,
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357 vlc_tab_fft_tone_offset_4_huffbits, 1, 1,
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358 vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
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359 }
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360
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361
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362 /* for floating point to fixed point conversion */
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363 static float f2i_scale = (float) (1 << (FRAC_BITS - 15));
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364
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365
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366 static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth)
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367 {
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368 int value;
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369
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370 value = get_vlc2(gb, vlc->table, vlc->bits, depth);
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371
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372 /* stage-2, 3 bits exponent escape sequence */
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373 if (value-- == 0)
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374 value = get_bits (gb, get_bits (gb, 3) + 1);
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375
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376 /* stage-3, optional */
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377 if (flag) {
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378 int tmp = vlc_stage3_values[value];
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379
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380 if ((value & ~3) > 0)
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381 tmp += get_bits (gb, (value >> 2));
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382 value = tmp;
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383 }
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384
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385 return value;
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386 }
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387
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388
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389 static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth)
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390 {
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391 int value = qdm2_get_vlc (gb, vlc, 0, depth);
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392
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393 return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
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394 }
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395
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396
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397 /**
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398 * QDM2 checksum
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399 *
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400 * @param data pointer to data to be checksum'ed
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401 * @param length data length
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402 * @param value checksum value
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403 *
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404 * @return 0 if checksum is ok
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405 */
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406 static uint16_t qdm2_packet_checksum (uint8_t *data, int length, int value) {
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407 int i;
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408
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409 for (i=0; i < length; i++)
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410 value -= data[i];
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411
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412 return (uint16_t)(value & 0xffff);
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413 }
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414
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415
|
|
416 /**
|
|
417 * Fills a QDM2SubPacket structure with packet type, size, and data pointer
|
|
418 *
|
|
419 * @param gb bitreader context
|
|
420 * @param sub_packet packet under analysis
|
|
421 */
|
|
422 static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub_packet)
|
|
423 {
|
|
424 sub_packet->type = get_bits (gb, 8);
|
|
425
|
|
426 if (sub_packet->type == 0) {
|
|
427 sub_packet->size = 0;
|
|
428 sub_packet->data = NULL;
|
|
429 } else {
|
|
430 sub_packet->size = get_bits (gb, 8);
|
|
431
|
|
432 if (sub_packet->type & 0x80) {
|
|
433 sub_packet->size <<= 8;
|
|
434 sub_packet->size |= get_bits (gb, 8);
|
|
435 sub_packet->type &= 0x7f;
|
|
436 }
|
|
437
|
|
438 if (sub_packet->type == 0x7f)
|
|
439 sub_packet->type |= (get_bits (gb, 8) << 8);
|
|
440
|
|
441 sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data
|
|
442 }
|
|
443
|
|
444 av_log(NULL,AV_LOG_DEBUG,"Sub packet: type=%d size=%d start_offs=%x\n",
|
|
445 sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
|
|
446 }
|
|
447
|
|
448
|
|
449 /**
|
|
450 * Return node pointer to first packet of requested type in list
|
|
451 *
|
|
452 * @param list list of subpacket to be scanned
|
|
453 * @param type type of searched subpacket
|
|
454 * @return node pointer for subpacket if found, else NULL
|
|
455 */
|
|
456 static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int type)
|
|
457 {
|
|
458 while (list != NULL && list->packet != NULL) {
|
|
459 if (list->packet->type == type)
|
|
460 return list;
|
|
461 list = list->next;
|
|
462 }
|
|
463 return NULL;
|
|
464 }
|
|
465
|
|
466
|
|
467 /**
|
|
468 * Replaces 8 elements with their average value
|
|
469 * Called by qdm2_decode_superblock before starting subblocks decoding
|
|
470 *
|
|
471 * @param q context
|
|
472 */
|
|
473 static void average_quantized_coeffs (QDM2Context *q)
|
|
474 {
|
|
475 int i, j, n, ch, sum;
|
|
476
|
|
477 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
|
|
478
|
|
479 for (ch = 0; ch < q->nb_channels; ch++)
|
|
480 for (i = 0; i < n; i++) {
|
|
481 sum = 0;
|
|
482
|
|
483 for (j = 0; j < 8; j++)
|
|
484 sum += q->quantized_coeffs[ch][i][j];
|
|
485
|
|
486 sum /= 8;
|
|
487 if (sum > 0)
|
|
488 sum--;
|
|
489
|
|
490 for (j=0; j < 8; j++)
|
|
491 q->quantized_coeffs[ch][i][j] = sum;
|
|
492 }
|
|
493 }
|
|
494
|
|
495
|
|
496 /**
|
|
497 * Build subband samples with noise weighted by q->tone_level
|
|
498 * Called by synthfilt_build_sb_samples
|
|
499 *
|
|
500 * @param q context
|
|
501 * @param sb subband index
|
|
502 */
|
|
503 static void build_sb_samples_from_noise (QDM2Context *q, int sb)
|
|
504 {
|
|
505 int ch, j;
|
|
506
|
|
507 FIX_NOISE_IDX(q->noise_idx);
|
|
508
|
|
509 if (!q->nb_channels)
|
|
510 return;
|
|
511
|
|
512 for (ch = 0; ch < q->nb_channels; ch++)
|
|
513 for (j = 0; j < 64; j++) {
|
|
514 q->sb_samples[ch][j * 2][sb] = (int32_t)(f2i_scale * SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j] + .5);
|
|
515 q->sb_samples[ch][j * 2 + 1][sb] = (int32_t)(f2i_scale * SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j] + .5);
|
|
516 }
|
|
517 }
|
|
518
|
|
519
|
|
520 /**
|
|
521 * Called while processing data from subpackets 11 and 12
|
|
522 * Used after making changes to coding_method array
|
|
523 *
|
|
524 * @param sb subband index
|
|
525 * @param channels number of channels
|
|
526 * @param coding_method q->coding_method[0][0][0]
|
|
527 */
|
|
528 void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method)
|
|
529 {
|
|
530 int j,k;
|
|
531 int ch;
|
|
532 int run, case_val;
|
|
533 int switchtable[23] = {0,5,1,5,5,5,5,5,2,5,5,5,5,5,5,5,3,5,5,5,5,5,4};
|
|
534
|
|
535 for (ch = 0; ch < channels; ch++) {
|
|
536 for (j = 0; j < 64; ) {
|
|
537 if((coding_method[ch][sb][j] - 8) > 22) {
|
|
538 run = 1;
|
|
539 case_val = 8;
|
|
540 } else {
|
|
541 switch (switchtable[coding_method[ch][sb][j]]) {
|
|
542 case 0: run = 10; case_val = 10; break;
|
|
543 case 1: run = 1; case_val = 16; break;
|
|
544 case 2: run = 5; case_val = 24; break;
|
|
545 case 3: run = 3; case_val = 30; break;
|
|
546 case 4: run = 1; case_val = 30; break;
|
|
547 case 5: run = 1; case_val = 8; break;
|
|
548 default: run = 1; case_val = 8; break;
|
|
549 }
|
|
550 }
|
|
551 for (k = 0; k < run; k++)
|
|
552 if (j + k < 128)
|
|
553 if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j])
|
|
554 if (k > 0) {
|
|
555 SAMPLES_NEEDED
|
|
556 //not debugged, almost never used
|
|
557 memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8_t));
|
|
558 memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8_t));
|
|
559 }
|
|
560 j += run;
|
|
561 }
|
|
562 }
|
|
563 }
|
|
564
|
|
565
|
|
566 /**
|
|
567 * Related to synthesis filter
|
|
568 * Called by process_subpacket_10
|
|
569 *
|
|
570 * @param q context
|
|
571 * @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10
|
|
572 */
|
|
573 static void fill_tone_level_array (QDM2Context *q, int flag)
|
|
574 {
|
|
575 int i, sb, ch, sb_used;
|
|
576 int tmp, tab;
|
|
577
|
|
578 // This should never happen
|
|
579 if (q->nb_channels <= 0)
|
|
580 return;
|
|
581
|
|
582 for (ch = 0; ch < q->nb_channels; ch++)
|
|
583 for (sb = 0; sb < 30; sb++)
|
|
584 for (i = 0; i < 8; i++) {
|
|
585 if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1))
|
|
586 tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+
|
|
587 q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
|
|
588 else
|
|
589 tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
|
|
590 if(tmp < 0)
|
|
591 tmp += 0xff;
|
|
592 q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff;
|
|
593 }
|
|
594
|
|
595 sb_used = QDM2_SB_USED(q->sub_sampling);
|
|
596
|
|
597 if ((q->superblocktype_2_3 != 0) && !flag) {
|
|
598 for (sb = 0; sb < sb_used; sb++)
|
|
599 for (ch = 0; ch < q->nb_channels; ch++)
|
|
600 for (i = 0; i < 64; i++) {
|
|
601 q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
|
|
602 if (q->tone_level_idx[ch][sb][i] < 0)
|
|
603 q->tone_level[ch][sb][i] = 0;
|
|
604 else
|
|
605 q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f];
|
|
606 }
|
|
607 } else {
|
|
608 tab = q->superblocktype_2_3 ? 0 : 1;
|
|
609 for (sb = 0; sb < sb_used; sb++) {
|
|
610 if ((sb >= 4) && (sb <= 23)) {
|
|
611 for (ch = 0; ch < q->nb_channels; ch++)
|
|
612 for (i = 0; i < 64; i++) {
|
|
613 tmp = q->tone_level_idx_base[ch][sb][i / 8] -
|
|
614 q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] -
|
|
615 q->tone_level_idx_mid[ch][sb - 4][i / 8] -
|
|
616 q->tone_level_idx_hi2[ch][sb - 4];
|
|
617 q->tone_level_idx[ch][sb][i] = tmp & 0xff;
|
|
618 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
|
|
619 q->tone_level[ch][sb][i] = 0;
|
|
620 else
|
|
621 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
|
|
622 }
|
|
623 } else {
|
|
624 if (sb > 4) {
|
|
625 for (ch = 0; ch < q->nb_channels; ch++)
|
|
626 for (i = 0; i < 64; i++) {
|
|
627 tmp = q->tone_level_idx_base[ch][sb][i / 8] -
|
|
628 q->tone_level_idx_hi1[ch][2][i / 8][i % 8] -
|
|
629 q->tone_level_idx_hi2[ch][sb - 4];
|
|
630 q->tone_level_idx[ch][sb][i] = tmp & 0xff;
|
|
631 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
|
|
632 q->tone_level[ch][sb][i] = 0;
|
|
633 else
|
|
634 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
|
|
635 }
|
|
636 } else {
|
|
637 for (ch = 0; ch < q->nb_channels; ch++)
|
|
638 for (i = 0; i < 64; i++) {
|
|
639 tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
|
|
640 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
|
|
641 q->tone_level[ch][sb][i] = 0;
|
|
642 else
|
|
643 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
|
|
644 }
|
|
645 }
|
|
646 }
|
|
647 }
|
|
648 }
|
|
649
|
|
650 return;
|
|
651 }
|
|
652
|
|
653
|
|
654 /**
|
|
655 * Related to synthesis filter
|
|
656 * Called by process_subpacket_11
|
|
657 * c is built with data from subpacket 11
|
|
658 * Most of this function is used only if superblock_type_2_3 == 0, never seen it in samples
|
|
659 *
|
2967
|
660 * @param tone_level_idx
|
2914
|
661 * @param tone_level_idx_temp
|
|
662 * @param coding_method q->coding_method[0][0][0]
|
|
663 * @param nb_channels number of channels
|
|
664 * @param c coming from subpacket 11, passed as 8*c
|
|
665 * @param superblocktype_2_3 flag based on superblock packet type
|
|
666 * @param cm_table_select q->cm_table_select
|
|
667 */
|
|
668 static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp,
|
|
669 sb_int8_array coding_method, int nb_channels,
|
|
670 int c, int superblocktype_2_3, int cm_table_select)
|
|
671 {
|
|
672 int ch, sb, j;
|
|
673 int tmp, acc, esp_40, comp;
|
|
674 int add1, add2, add3, add4;
|
|
675 int64_t multres;
|
|
676
|
|
677 // This should never happen
|
|
678 if (nb_channels <= 0)
|
|
679 return;
|
|
680
|
|
681 if (!superblocktype_2_3) {
|
|
682 /* This case is untested, no samples available */
|
|
683 SAMPLES_NEEDED
|
|
684 for (ch = 0; ch < nb_channels; ch++)
|
|
685 for (sb = 0; sb < 30; sb++) {
|
|
686 for (j = 1; j < 64; j++) {
|
|
687 add1 = tone_level_idx[ch][sb][j] - 10;
|
|
688 if (add1 < 0)
|
|
689 add1 = 0;
|
|
690 add2 = add3 = add4 = 0;
|
|
691 if (sb > 1) {
|
|
692 add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
|
|
693 if (add2 < 0)
|
|
694 add2 = 0;
|
|
695 }
|
|
696 if (sb > 0) {
|
|
697 add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
|
|
698 if (add3 < 0)
|
|
699 add3 = 0;
|
|
700 }
|
|
701 if (sb < 29) {
|
|
702 add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
|
|
703 if (add4 < 0)
|
|
704 add4 = 0;
|
|
705 }
|
|
706 tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
|
|
707 if (tmp < 0)
|
|
708 tmp = 0;
|
|
709 tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
|
|
710 }
|
|
711 tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
|
|
712 }
|
|
713 acc = 0;
|
|
714 for (ch = 0; ch < nb_channels; ch++)
|
|
715 for (sb = 0; sb < 30; sb++)
|
|
716 for (j = 0; j < 64; j++)
|
|
717 acc += tone_level_idx_temp[ch][sb][j];
|
|
718 if (acc)
|
|
719 tmp = c * 256 / (acc & 0xffff);
|
|
720 multres = 0x66666667 * (acc * 10);
|
|
721 esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
|
|
722 for (ch = 0; ch < nb_channels; ch++)
|
|
723 for (sb = 0; sb < 30; sb++)
|
|
724 for (j = 0; j < 64; j++) {
|
|
725 comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
|
|
726 if (comp < 0)
|
|
727 comp += 0xff;
|
|
728 comp /= 256; // signed shift
|
|
729 switch(sb) {
|
|
730 case 0:
|
|
731 if (comp < 30)
|
|
732 comp = 30;
|
|
733 comp += 15;
|
|
734 break;
|
|
735 case 1:
|
|
736 if (comp < 24)
|
|
737 comp = 24;
|
|
738 comp += 10;
|
|
739 break;
|
|
740 case 2:
|
|
741 case 3:
|
|
742 case 4:
|
|
743 if (comp < 16)
|
|
744 comp = 16;
|
|
745 }
|
|
746 if (comp <= 5)
|
|
747 tmp = 0;
|
|
748 else if (comp <= 10)
|
|
749 tmp = 10;
|
|
750 else if (comp <= 16)
|
|
751 tmp = 16;
|
|
752 else if (comp <= 24)
|
|
753 tmp = -1;
|
|
754 else
|
|
755 tmp = 0;
|
|
756 coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
|
|
757 }
|
|
758 for (sb = 0; sb < 30; sb++)
|
|
759 fix_coding_method_array(sb, nb_channels, coding_method);
|
|
760 for (ch = 0; ch < nb_channels; ch++)
|
|
761 for (sb = 0; sb < 30; sb++)
|
|
762 for (j = 0; j < 64; j++)
|
|
763 if (sb >= 10) {
|
|
764 if (coding_method[ch][sb][j] < 10)
|
|
765 coding_method[ch][sb][j] = 10;
|
|
766 } else {
|
|
767 if (sb >= 2) {
|
|
768 if (coding_method[ch][sb][j] < 16)
|
|
769 coding_method[ch][sb][j] = 16;
|
|
770 } else {
|
|
771 if (coding_method[ch][sb][j] < 30)
|
|
772 coding_method[ch][sb][j] = 30;
|
|
773 }
|
|
774 }
|
|
775 } else { // superblocktype_2_3 != 0
|
|
776 for (ch = 0; ch < nb_channels; ch++)
|
|
777 for (sb = 0; sb < 30; sb++)
|
|
778 for (j = 0; j < 64; j++)
|
|
779 coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
|
|
780 }
|
|
781
|
|
782 return;
|
|
783 }
|
|
784
|
|
785
|
|
786 /**
|
|
787 *
|
|
788 * Called by process_subpacket_11 to process more data from subpacket 11 with sb 0-8
|
|
789 * Called by process_subpacket_12 to process data from subpacket 12 with sb 8-sb_used
|
|
790 *
|
|
791 * @param q context
|
|
792 * @param gb bitreader context
|
|
793 * @param length packet length in bit
|
|
794 * @param sb_min lower subband processed (sb_min included)
|
|
795 * @param sb_max higher subband processed (sb_max excluded)
|
|
796 */
|
|
797 static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max)
|
|
798 {
|
|
799 int sb, j, k, n, ch, run, channels;
|
|
800 int joined_stereo, zero_encoding, chs;
|
|
801 int type34_first;
|
|
802 float type34_div = 0;
|
|
803 float type34_predictor;
|
|
804 float samples[10], sign_bits[16];
|
|
805
|
|
806 if (length == 0) {
|
|
807 // If no data use noise
|
|
808 for (sb=sb_min; sb < sb_max; sb++)
|
|
809 build_sb_samples_from_noise (q, sb);
|
|
810
|
|
811 return;
|
|
812 }
|
|
813
|
|
814 for (sb = sb_min; sb < sb_max; sb++) {
|
|
815 FIX_NOISE_IDX(q->noise_idx);
|
|
816
|
|
817 channels = q->nb_channels;
|
|
818
|
|
819 if (q->nb_channels <= 1 || sb < 12)
|
|
820 joined_stereo = 0;
|
|
821 else if (sb >= 24)
|
|
822 joined_stereo = 1;
|
|
823 else
|
|
824 joined_stereo = (BITS_LEFT(length,gb) >= 1) ? get_bits1 (gb) : 0;
|
|
825
|
|
826 if (joined_stereo) {
|
|
827 if (BITS_LEFT(length,gb) >= 16)
|
|
828 for (j = 0; j < 16; j++)
|
|
829 sign_bits[j] = get_bits1 (gb);
|
|
830
|
|
831 for (j = 0; j < 64; j++)
|
|
832 if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
|
|
833 q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
|
|
834
|
|
835 fix_coding_method_array(sb, q->nb_channels, q->coding_method);
|
|
836 channels = 1;
|
|
837 }
|
|
838
|
|
839 for (ch = 0; ch < channels; ch++) {
|
|
840 zero_encoding = (BITS_LEFT(length,gb) >= 1) ? get_bits1(gb) : 0;
|
|
841 type34_predictor = 0.0;
|
|
842 type34_first = 1;
|
|
843
|
|
844 for (j = 0; j < 128; ) {
|
|
845 switch (q->coding_method[ch][sb][j / 2]) {
|
|
846 case 8:
|
|
847 if (BITS_LEFT(length,gb) >= 10) {
|
|
848 if (zero_encoding) {
|
|
849 for (k = 0; k < 5; k++) {
|
|
850 if ((j + 2 * k) >= 128)
|
|
851 break;
|
|
852 samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
|
|
853 }
|
|
854 } else {
|
|
855 n = get_bits(gb, 8);
|
|
856 for (k = 0; k < 5; k++)
|
|
857 samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
|
|
858 }
|
|
859 for (k = 0; k < 5; k++)
|
|
860 samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
|
|
861 } else {
|
|
862 for (k = 0; k < 10; k++)
|
|
863 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
|
|
864 }
|
|
865 run = 10;
|
|
866 break;
|
|
867
|
|
868 case 10:
|
|
869 if (BITS_LEFT(length,gb) >= 1) {
|
|
870 float f = 0.81;
|
|
871
|
|
872 if (get_bits1(gb))
|
|
873 f = -f;
|
|
874 f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
|
|
875 samples[0] = f;
|
|
876 } else {
|
|
877 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
|
|
878 }
|
|
879 run = 1;
|
|
880 break;
|
|
881
|
|
882 case 16:
|
|
883 if (BITS_LEFT(length,gb) >= 10) {
|
|
884 if (zero_encoding) {
|
|
885 for (k = 0; k < 5; k++) {
|
|
886 if ((j + k) >= 128)
|
|
887 break;
|
|
888 samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
|
|
889 }
|
|
890 } else {
|
|
891 n = get_bits (gb, 8);
|
|
892 for (k = 0; k < 5; k++)
|
|
893 samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
|
|
894 }
|
|
895 } else {
|
|
896 for (k = 0; k < 5; k++)
|
|
897 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
|
|
898 }
|
|
899 run = 5;
|
|
900 break;
|
|
901
|
|
902 case 24:
|
|
903 if (BITS_LEFT(length,gb) >= 7) {
|
|
904 n = get_bits(gb, 7);
|
|
905 for (k = 0; k < 3; k++)
|
|
906 samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
|
|
907 } else {
|
|
908 for (k = 0; k < 3; k++)
|
|
909 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
|
|
910 }
|
|
911 run = 3;
|
|
912 break;
|
|
913
|
|
914 case 30:
|
|
915 if (BITS_LEFT(length,gb) >= 4)
|
|
916 samples[0] = type30_dequant[qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1)];
|
|
917 else
|
|
918 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
|
2967
|
919
|
2914
|
920 run = 1;
|
|
921 break;
|
|
922
|
|
923 case 34:
|
|
924 if (BITS_LEFT(length,gb) >= 7) {
|
|
925 if (type34_first) {
|
|
926 type34_div = (float)(1 << get_bits(gb, 2));
|
|
927 samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
|
|
928 type34_predictor = samples[0];
|
|
929 type34_first = 0;
|
|
930 } else {
|
|
931 samples[0] = type34_delta[qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1)] / type34_div + type34_predictor;
|
|
932 type34_predictor = samples[0];
|
|
933 }
|
|
934 } else {
|
|
935 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
|
|
936 }
|
|
937 run = 1;
|
|
938 break;
|
|
939
|
|
940 default:
|
|
941 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
|
|
942 run = 1;
|
|
943 break;
|
|
944 }
|
|
945
|
|
946 if (joined_stereo) {
|
|
947 float tmp[10][MPA_MAX_CHANNELS];
|
|
948
|
|
949 for (k = 0; k < run; k++) {
|
|
950 tmp[k][0] = samples[k];
|
|
951 tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k];
|
|
952 }
|
|
953 for (chs = 0; chs < q->nb_channels; chs++)
|
|
954 for (k = 0; k < run; k++)
|
|
955 if ((j + k) < 128)
|
|
956 q->sb_samples[chs][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs] + .5);
|
|
957 } else {
|
|
958 for (k = 0; k < run; k++)
|
|
959 if ((j + k) < 128)
|
|
960 q->sb_samples[ch][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[ch][sb][(j + k)/2] * samples[k] + .5);
|
|
961 }
|
|
962
|
|
963 j += run;
|
|
964 } // j loop
|
|
965 } // channel loop
|
|
966 } // subband loop
|
|
967 }
|
|
968
|
|
969
|
|
970 /**
|
|
971 * Init the first element of a channel in quantized_coeffs with data from packet 10 (quantized_coeffs[ch][0])
|
|
972 * This is similar to process_subpacket_9, but for a single channel and for element [0]
|
|
973 * same VLC tables as process_subpacket_9 are used
|
|
974 *
|
|
975 * @param q context
|
|
976 * @param quantized_coeffs pointer to quantized_coeffs[ch][0]
|
|
977 * @param gb bitreader context
|
|
978 * @param length packet length in bit
|
|
979 */
|
|
980 static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb, int length)
|
|
981 {
|
|
982 int i, k, run, level, diff;
|
|
983
|
|
984 if (BITS_LEFT(length,gb) < 16)
|
|
985 return;
|
|
986 level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
|
|
987
|
|
988 quantized_coeffs[0] = level;
|
|
989
|
|
990 for (i = 0; i < 7; ) {
|
|
991 if (BITS_LEFT(length,gb) < 16)
|
|
992 break;
|
|
993 run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
|
|
994
|
|
995 if (BITS_LEFT(length,gb) < 16)
|
|
996 break;
|
|
997 diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
|
2967
|
998
|
2914
|
999 for (k = 1; k <= run; k++)
|
|
1000 quantized_coeffs[i + k] = (level + ((k * diff) / run));
|
2967
|
1001
|
2914
|
1002 level += diff;
|
|
1003 i += run;
|
|
1004 }
|
|
1005 }
|
|
1006
|
|
1007
|
|
1008 /**
|
|
1009 * Related to synthesis filter, process data from packet 10
|
|
1010 * Init part of quantized_coeffs via function init_quantized_coeffs_elem0
|
|
1011 * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with data from packet 10
|
|
1012 *
|
|
1013 * @param q context
|
|
1014 * @param gb bitreader context
|
|
1015 * @param length packet length in bit
|
|
1016 */
|
|
1017 static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, int length)
|
|
1018 {
|
|
1019 int sb, j, k, n, ch;
|
|
1020
|
|
1021 for (ch = 0; ch < q->nb_channels; ch++) {
|
|
1022 init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb, length);
|
|
1023
|
|
1024 if (BITS_LEFT(length,gb) < 16) {
|
|
1025 memset(q->quantized_coeffs[ch][0], 0, 8);
|
|
1026 break;
|
|
1027 }
|
|
1028 }
|
|
1029
|
|
1030 n = q->sub_sampling + 1;
|
|
1031
|
|
1032 for (sb = 0; sb < n; sb++)
|
|
1033 for (ch = 0; ch < q->nb_channels; ch++)
|
|
1034 for (j = 0; j < 8; j++) {
|
|
1035 if (BITS_LEFT(length,gb) < 1)
|
|
1036 break;
|
|
1037 if (get_bits1(gb)) {
|
|
1038 for (k=0; k < 8; k++) {
|
|
1039 if (BITS_LEFT(length,gb) < 16)
|
|
1040 break;
|
|
1041 q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
|
|
1042 }
|
|
1043 } else {
|
|
1044 for (k=0; k < 8; k++)
|
|
1045 q->tone_level_idx_hi1[ch][sb][j][k] = 0;
|
|
1046 }
|
|
1047 }
|
|
1048
|
|
1049 n = QDM2_SB_USED(q->sub_sampling) - 4;
|
|
1050
|
|
1051 for (sb = 0; sb < n; sb++)
|
|
1052 for (ch = 0; ch < q->nb_channels; ch++) {
|
|
1053 if (BITS_LEFT(length,gb) < 16)
|
|
1054 break;
|
|
1055 q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
|
|
1056 if (sb > 19)
|
|
1057 q->tone_level_idx_hi2[ch][sb] -= 16;
|
|
1058 else
|
|
1059 for (j = 0; j < 8; j++)
|
|
1060 q->tone_level_idx_mid[ch][sb][j] = -16;
|
|
1061 }
|
|
1062
|
|
1063 n = QDM2_SB_USED(q->sub_sampling) - 5;
|
|
1064
|
|
1065 for (sb = 0; sb < n; sb++)
|
|
1066 for (ch = 0; ch < q->nb_channels; ch++)
|
|
1067 for (j = 0; j < 8; j++) {
|
|
1068 if (BITS_LEFT(length,gb) < 16)
|
|
1069 break;
|
|
1070 q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
|
|
1071 }
|
|
1072 }
|
|
1073
|
|
1074 /**
|
|
1075 * Process subpacket 9, init quantized_coeffs with data from it
|
|
1076 *
|
|
1077 * @param q context
|
|
1078 * @param node pointer to node with packet
|
|
1079 */
|
|
1080 static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node)
|
|
1081 {
|
|
1082 GetBitContext gb;
|
|
1083 int i, j, k, n, ch, run, level, diff;
|
|
1084
|
2916
|
1085 init_get_bits(&gb, node->packet->data, node->packet->size*8);
|
2914
|
1086
|
|
1087 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function
|
|
1088
|
|
1089 for (i = 1; i < n; i++)
|
|
1090 for (ch=0; ch < q->nb_channels; ch++) {
|
|
1091 level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
|
|
1092 q->quantized_coeffs[ch][i][0] = level;
|
|
1093
|
|
1094 for (j = 0; j < (8 - 1); ) {
|
|
1095 run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
|
|
1096 diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
|
|
1097
|
|
1098 for (k = 1; k <= run; k++)
|
|
1099 q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run));
|
|
1100
|
|
1101 level += diff;
|
|
1102 j += run;
|
|
1103 }
|
|
1104 }
|
|
1105
|
|
1106 for (ch = 0; ch < q->nb_channels; ch++)
|
|
1107 for (i = 0; i < 8; i++)
|
|
1108 q->quantized_coeffs[ch][0][i] = 0;
|
|
1109 }
|
|
1110
|
|
1111
|
|
1112 /**
|
|
1113 * Process subpacket 10 if not null, else
|
|
1114 *
|
|
1115 * @param q context
|
|
1116 * @param node pointer to node with packet
|
|
1117 * @param length packet length in bit
|
|
1118 */
|
|
1119 static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node, int length)
|
|
1120 {
|
|
1121 GetBitContext gb;
|
|
1122
|
2916
|
1123 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
|
2914
|
1124
|
|
1125 if (length != 0) {
|
|
1126 init_tone_level_dequantization(q, &gb, length);
|
|
1127 fill_tone_level_array(q, 1);
|
|
1128 } else {
|
|
1129 fill_tone_level_array(q, 0);
|
|
1130 }
|
|
1131 }
|
|
1132
|
|
1133
|
|
1134 /**
|
|
1135 * Process subpacket 11
|
|
1136 *
|
|
1137 * @param q context
|
|
1138 * @param node pointer to node with packet
|
|
1139 * @param length packet length in bit
|
|
1140 */
|
|
1141 static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node, int length)
|
|
1142 {
|
|
1143 GetBitContext gb;
|
|
1144
|
2916
|
1145 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
|
2914
|
1146 if (length >= 32) {
|
|
1147 int c = get_bits (&gb, 13);
|
|
1148
|
|
1149 if (c > 3)
|
|
1150 fill_coding_method_array (q->tone_level_idx, q->tone_level_idx_temp, q->coding_method,
|
|
1151 q->nb_channels, 8*c, q->superblocktype_2_3, q->cm_table_select);
|
|
1152 }
|
|
1153
|
|
1154 synthfilt_build_sb_samples(q, &gb, length, 0, 8);
|
|
1155 }
|
|
1156
|
|
1157
|
|
1158 /**
|
|
1159 * Process subpacket 12
|
|
1160 *
|
|
1161 * @param q context
|
|
1162 * @param node pointer to node with packet
|
|
1163 * @param length packet length in bit
|
|
1164 */
|
|
1165 static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node, int length)
|
|
1166 {
|
|
1167 GetBitContext gb;
|
|
1168
|
2916
|
1169 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
|
2914
|
1170 synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
|
|
1171 }
|
|
1172
|
|
1173 /*
|
|
1174 * Process new subpackets for synthesis filter
|
|
1175 *
|
|
1176 * @param q context
|
|
1177 * @param list list with synthesis filter packets (list D)
|
|
1178 */
|
|
1179 static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list)
|
|
1180 {
|
|
1181 QDM2SubPNode *nodes[4];
|
|
1182
|
|
1183 nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
|
|
1184 if (nodes[0] != NULL)
|
|
1185 process_subpacket_9(q, nodes[0]);
|
|
1186
|
|
1187 nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
|
|
1188 if (nodes[1] != NULL)
|
|
1189 process_subpacket_10(q, nodes[1], nodes[1]->packet->size << 3);
|
|
1190 else
|
|
1191 process_subpacket_10(q, NULL, 0);
|
|
1192
|
|
1193 nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
|
|
1194 if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL)
|
|
1195 process_subpacket_11(q, nodes[2], (nodes[2]->packet->size << 3));
|
|
1196 else
|
|
1197 process_subpacket_11(q, NULL, 0);
|
|
1198
|
|
1199 nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
|
|
1200 if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL)
|
|
1201 process_subpacket_12(q, nodes[3], (nodes[3]->packet->size << 3));
|
|
1202 else
|
|
1203 process_subpacket_12(q, NULL, 0);
|
|
1204 }
|
|
1205
|
|
1206
|
|
1207 /*
|
|
1208 * Decode superblock, fill packet lists
|
|
1209 *
|
|
1210 * @param q context
|
|
1211 */
|
|
1212 static void qdm2_decode_super_block (QDM2Context *q)
|
|
1213 {
|
|
1214 GetBitContext gb;
|
|
1215 QDM2SubPacket header, *packet;
|
|
1216 int i, packet_bytes, sub_packet_size, sub_packets_D;
|
|
1217 unsigned int next_index = 0;
|
|
1218
|
|
1219 memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1));
|
|
1220 memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid));
|
|
1221 memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
|
|
1222
|
|
1223 q->sub_packets_B = 0;
|
|
1224 sub_packets_D = 0;
|
|
1225
|
|
1226 average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
|
|
1227
|
2916
|
1228 init_get_bits(&gb, q->compressed_data, q->compressed_size*8);
|
2914
|
1229 qdm2_decode_sub_packet_header(&gb, &header);
|
|
1230
|
|
1231 if (header.type < 2 || header.type >= 8) {
|
|
1232 q->has_errors = 1;
|
|
1233 av_log(NULL,AV_LOG_ERROR,"bad superblock type\n");
|
|
1234 return;
|
|
1235 }
|
|
1236
|
|
1237 q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
|
|
1238 packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
|
|
1239
|
2916
|
1240 init_get_bits(&gb, header.data, header.size*8);
|
2914
|
1241
|
|
1242 if (header.type == 2 || header.type == 4 || header.type == 5) {
|
|
1243 int csum = 257 * get_bits(&gb, 8) + 2 * get_bits(&gb, 8);
|
|
1244
|
|
1245 csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
|
|
1246
|
|
1247 if (csum != 0) {
|
|
1248 q->has_errors = 1;
|
|
1249 av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n");
|
|
1250 return;
|
|
1251 }
|
|
1252 }
|
|
1253
|
|
1254 q->sub_packet_list_B[0].packet = NULL;
|
|
1255 q->sub_packet_list_D[0].packet = NULL;
|
|
1256
|
|
1257 for (i = 0; i < 6; i++)
|
|
1258 if (--q->fft_level_exp[i] < 0)
|
|
1259 q->fft_level_exp[i] = 0;
|
|
1260
|
|
1261 for (i = 0; packet_bytes > 0; i++) {
|
|
1262 int j;
|
|
1263
|
|
1264 q->sub_packet_list_A[i].next = NULL;
|
|
1265
|
|
1266 if (i > 0) {
|
|
1267 q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
|
|
1268
|
|
1269 /* seek to next block */
|
2916
|
1270 init_get_bits(&gb, header.data, header.size*8);
|
2914
|
1271 skip_bits(&gb, next_index*8);
|
|
1272
|
|
1273 if (next_index >= header.size)
|
|
1274 break;
|
|
1275 }
|
|
1276
|
|
1277 /* decode sub packet */
|
|
1278 packet = &q->sub_packets[i];
|
|
1279 qdm2_decode_sub_packet_header(&gb, packet);
|
|
1280 next_index = packet->size + get_bits_count(&gb) / 8;
|
|
1281 sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
|
|
1282
|
|
1283 if (packet->type == 0)
|
|
1284 break;
|
|
1285
|
|
1286 if (sub_packet_size > packet_bytes) {
|
|
1287 if (packet->type != 10 && packet->type != 11 && packet->type != 12)
|
|
1288 break;
|
|
1289 packet->size += packet_bytes - sub_packet_size;
|
|
1290 }
|
|
1291
|
|
1292 packet_bytes -= sub_packet_size;
|
|
1293
|
|
1294 /* add sub packet to 'all sub packets' list */
|
|
1295 q->sub_packet_list_A[i].packet = packet;
|
|
1296
|
|
1297 /* add sub packet to related list */
|
|
1298 if (packet->type == 8) {
|
|
1299 SAMPLES_NEEDED_2("packet type 8");
|
|
1300 return;
|
|
1301 } else if (packet->type >= 9 && packet->type <= 12) {
|
|
1302 /* packets for MPEG Audio like Synthesis Filter */
|
|
1303 QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
|
|
1304 } else if (packet->type == 13) {
|
|
1305 for (j = 0; j < 6; j++)
|
|
1306 q->fft_level_exp[j] = get_bits(&gb, 6);
|
|
1307 } else if (packet->type == 14) {
|
|
1308 for (j = 0; j < 6; j++)
|
|
1309 q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
|
|
1310 } else if (packet->type == 15) {
|
|
1311 SAMPLES_NEEDED_2("packet type 15")
|
|
1312 return;
|
|
1313 } else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) {
|
|
1314 /* packets for FFT */
|
|
1315 QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet);
|
|
1316 }
|
|
1317 } // Packet bytes loop
|
|
1318
|
|
1319 /* **************************************************************** */
|
|
1320 if (q->sub_packet_list_D[0].packet != NULL) {
|
|
1321 process_synthesis_subpackets(q, q->sub_packet_list_D);
|
|
1322 q->do_synth_filter = 1;
|
|
1323 } else if (q->do_synth_filter) {
|
|
1324 process_subpacket_10(q, NULL, 0);
|
|
1325 process_subpacket_11(q, NULL, 0);
|
|
1326 process_subpacket_12(q, NULL, 0);
|
|
1327 }
|
|
1328 /* **************************************************************** */
|
|
1329 }
|
|
1330
|
|
1331
|
|
1332 static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet,
|
|
1333 int offset, int duration, int channel,
|
|
1334 int exp, int phase)
|
|
1335 {
|
|
1336 if (q->fft_coefs_min_index[duration] < 0)
|
|
1337 q->fft_coefs_min_index[duration] = q->fft_coefs_index;
|
|
1338
|
|
1339 q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
|
|
1340 q->fft_coefs[q->fft_coefs_index].channel = channel;
|
|
1341 q->fft_coefs[q->fft_coefs_index].offset = offset;
|
|
1342 q->fft_coefs[q->fft_coefs_index].exp = exp;
|
|
1343 q->fft_coefs[q->fft_coefs_index].phase = phase;
|
|
1344 q->fft_coefs_index++;
|
|
1345 }
|
|
1346
|
|
1347
|
|
1348 static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *gb, int b)
|
|
1349 {
|
|
1350 int channel, stereo, phase, exp;
|
|
1351 int local_int_4, local_int_8, stereo_phase, local_int_10;
|
|
1352 int local_int_14, stereo_exp, local_int_20, local_int_28;
|
|
1353 int n, offset;
|
|
1354
|
|
1355 local_int_4 = 0;
|
|
1356 local_int_28 = 0;
|
|
1357 local_int_20 = 2;
|
|
1358 local_int_8 = (4 - duration);
|
|
1359 local_int_10 = 1 << (q->group_order - duration - 1);
|
|
1360 offset = 1;
|
|
1361
|
|
1362 while (1) {
|
|
1363 if (q->superblocktype_2_3) {
|
|
1364 while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
|
|
1365 offset = 1;
|
|
1366 if (n == 0) {
|
|
1367 local_int_4 += local_int_10;
|
|
1368 local_int_28 += (1 << local_int_8);
|
|
1369 } else {
|
|
1370 local_int_4 += 8*local_int_10;
|
|
1371 local_int_28 += (8 << local_int_8);
|
|
1372 }
|
|
1373 }
|
|
1374 offset += (n - 2);
|
|
1375 } else {
|
|
1376 offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
|
|
1377 while (offset >= (local_int_10 - 1)) {
|
|
1378 offset += (1 - (local_int_10 - 1));
|
|
1379 local_int_4 += local_int_10;
|
|
1380 local_int_28 += (1 << local_int_8);
|
|
1381 }
|
|
1382 }
|
|
1383
|
|
1384 if (local_int_4 >= q->group_size)
|
|
1385 return;
|
|
1386
|
|
1387 local_int_14 = (offset >> local_int_8);
|
|
1388
|
|
1389 if (q->nb_channels > 1) {
|
|
1390 channel = get_bits1(gb);
|
|
1391 stereo = get_bits1(gb);
|
|
1392 } else {
|
|
1393 channel = 0;
|
|
1394 stereo = 0;
|
|
1395 }
|
|
1396
|
|
1397 exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
|
|
1398 exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
|
|
1399 exp = (exp < 0) ? 0 : exp;
|
|
1400
|
|
1401 phase = get_bits(gb, 3);
|
|
1402 stereo_exp = 0;
|
|
1403 stereo_phase = 0;
|
|
1404
|
|
1405 if (stereo) {
|
|
1406 stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
|
|
1407 stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
|
|
1408 if (stereo_phase < 0)
|
|
1409 stereo_phase += 8;
|
|
1410 }
|
|
1411
|
|
1412 if (q->frequency_range > (local_int_14 + 1)) {
|
|
1413 int sub_packet = (local_int_20 + local_int_28);
|
|
1414
|
|
1415 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase);
|
|
1416 if (stereo)
|
|
1417 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase);
|
|
1418 }
|
|
1419
|
|
1420 offset++;
|
|
1421 }
|
|
1422 }
|
|
1423
|
|
1424
|
|
1425 static void qdm2_decode_fft_packets (QDM2Context *q)
|
|
1426 {
|
|
1427 int i, j, min, max, value, type, unknown_flag;
|
|
1428 GetBitContext gb;
|
|
1429
|
|
1430 if (q->sub_packet_list_B[0].packet == NULL)
|
|
1431 return;
|
|
1432
|
|
1433 /* reset minimum indices for FFT coefficients */
|
|
1434 q->fft_coefs_index = 0;
|
|
1435 for (i=0; i < 5; i++)
|
|
1436 q->fft_coefs_min_index[i] = -1;
|
|
1437
|
|
1438 /* process sub packets ordered by type, largest type first */
|
|
1439 for (i = 0, max = 256; i < q->sub_packets_B; i++) {
|
|
1440 QDM2SubPacket *packet;
|
|
1441
|
|
1442 /* find sub packet with largest type less than max */
|
|
1443 for (j = 0, min = 0, packet = NULL; j < q->sub_packets_B; j++) {
|
|
1444 value = q->sub_packet_list_B[j].packet->type;
|
|
1445 if (value > min && value < max) {
|
|
1446 min = value;
|
|
1447 packet = q->sub_packet_list_B[j].packet;
|
|
1448 }
|
|
1449 }
|
|
1450
|
|
1451 max = min;
|
|
1452
|
|
1453 /* check for errors (?) */
|
|
1454 if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16]))
|
|
1455 return;
|
|
1456
|
|
1457 /* decode FFT tones */
|
2916
|
1458 init_get_bits (&gb, packet->data, packet->size*8);
|
2914
|
1459
|
|
1460 if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
|
|
1461 unknown_flag = 1;
|
|
1462 else
|
|
1463 unknown_flag = 0;
|
|
1464
|
|
1465 type = packet->type;
|
|
1466
|
|
1467 if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
|
|
1468 int duration = q->sub_sampling + 5 - (type & 15);
|
|
1469
|
|
1470 if (duration >= 0 && duration < 4)
|
|
1471 qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
|
|
1472 } else if (type == 31) {
|
|
1473 for (i=0; i < 4; i++)
|
|
1474 qdm2_fft_decode_tones(q, i, &gb, unknown_flag);
|
|
1475 } else if (type == 46) {
|
|
1476 for (i=0; i < 6; i++)
|
|
1477 q->fft_level_exp[i] = get_bits(&gb, 6);
|
|
1478 for (i=0; i < 4; i++)
|
|
1479 qdm2_fft_decode_tones(q, i, &gb, unknown_flag);
|
|
1480 }
|
|
1481 } // Loop on B packets
|
|
1482
|
|
1483 /* calculate maximum indices for FFT coefficients */
|
|
1484 for (i = 0, j = -1; i < 5; i++)
|
|
1485 if (q->fft_coefs_min_index[i] >= 0) {
|
|
1486 if (j >= 0)
|
|
1487 q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i];
|
|
1488 j = i;
|
|
1489 }
|
|
1490 if (j >= 0)
|
|
1491 q->fft_coefs_max_index[j] = q->fft_coefs_index;
|
|
1492 }
|
|
1493
|
|
1494
|
|
1495 static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone)
|
|
1496 {
|
|
1497 float level, f[6];
|
|
1498 int i;
|
|
1499 QDM2Complex c;
|
|
1500 const double iscale = 2.0*M_PI / 512.0;
|
|
1501
|
|
1502 tone->phase += tone->phase_shift;
|
|
1503
|
|
1504 /* calculate current level (maximum amplitude) of tone */
|
|
1505 level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
|
|
1506 c.im = level * sin(tone->phase*iscale);
|
|
1507 c.re = level * cos(tone->phase*iscale);
|
|
1508
|
|
1509 /* generate FFT coefficients for tone */
|
|
1510 if (tone->duration >= 3 || tone->cutoff >= 3) {
|
|
1511 tone->samples_im[0] += c.im;
|
|
1512 tone->samples_re[0] += c.re;
|
|
1513 tone->samples_im[1] -= c.im;
|
|
1514 tone->samples_re[1] -= c.re;
|
|
1515 } else {
|
|
1516 f[1] = -tone->table[4];
|
|
1517 f[0] = tone->table[3] - tone->table[0];
|
|
1518 f[2] = 1.0 - tone->table[2] - tone->table[3];
|
|
1519 f[3] = tone->table[1] + tone->table[4] - 1.0;
|
|
1520 f[4] = tone->table[0] - tone->table[1];
|
|
1521 f[5] = tone->table[2];
|
|
1522 for (i = 0; i < 2; i++) {
|
|
1523 tone->samples_re[fft_cutoff_index_table[tone->cutoff][i]] += c.re * f[i];
|
|
1524 tone->samples_im[fft_cutoff_index_table[tone->cutoff][i]] += c.im *((tone->cutoff <= i) ? -f[i] : f[i]);
|
|
1525 }
|
|
1526 for (i = 0; i < 4; i++) {
|
|
1527 tone->samples_re[i] += c.re * f[i+2];
|
|
1528 tone->samples_im[i] += c.im * f[i+2];
|
|
1529 }
|
|
1530 }
|
|
1531
|
|
1532 /* copy the tone if it has not yet died out */
|
|
1533 if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
|
|
1534 memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
|
|
1535 q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
|
|
1536 }
|
|
1537 }
|
|
1538
|
|
1539
|
|
1540 static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet)
|
|
1541 {
|
|
1542 int i, j, ch;
|
|
1543 const double iscale = 0.25 * M_PI;
|
|
1544
|
|
1545 for (ch = 0; ch < q->channels; ch++) {
|
|
1546 memset(q->fft.samples_im[ch], 0, q->fft_size * sizeof(float));
|
|
1547 memset(q->fft.samples_re[ch], 0, q->fft_size * sizeof(float));
|
|
1548 }
|
|
1549
|
|
1550
|
|
1551 /* apply FFT tones with duration 4 (1 FFT period) */
|
|
1552 if (q->fft_coefs_min_index[4] >= 0)
|
|
1553 for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
|
|
1554 float level;
|
|
1555 QDM2Complex c;
|
|
1556
|
|
1557 if (q->fft_coefs[i].sub_packet != sub_packet)
|
|
1558 break;
|
|
1559
|
|
1560 ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel;
|
|
1561 level = (q->fft_coefs[i].exp < 0) ? 0.0 : fft_tone_level_table[q->superblocktype_2_3 ? 0 : 1][q->fft_coefs[i].exp & 63];
|
|
1562
|
|
1563 c.re = level * cos(q->fft_coefs[i].phase * iscale);
|
|
1564 c.im = level * sin(q->fft_coefs[i].phase * iscale);
|
|
1565 q->fft.samples_re[ch][q->fft_coefs[i].offset + 0] += c.re;
|
|
1566 q->fft.samples_im[ch][q->fft_coefs[i].offset + 0] += c.im;
|
|
1567 q->fft.samples_re[ch][q->fft_coefs[i].offset + 1] -= c.re;
|
|
1568 q->fft.samples_im[ch][q->fft_coefs[i].offset + 1] -= c.im;
|
|
1569 }
|
|
1570
|
|
1571 /* generate existing FFT tones */
|
|
1572 for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
|
|
1573 qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]);
|
|
1574 q->fft_tone_start = (q->fft_tone_start + 1) % 1000;
|
|
1575 }
|
|
1576
|
|
1577 /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
|
|
1578 for (i = 0; i < 4; i++)
|
|
1579 if (q->fft_coefs_min_index[i] >= 0) {
|
|
1580 for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
|
|
1581 int offset, four_i;
|
|
1582 FFTTone tone;
|
|
1583
|
|
1584 if (q->fft_coefs[j].sub_packet != sub_packet)
|
|
1585 break;
|
|
1586
|
|
1587 four_i = (4 - i);
|
|
1588 offset = q->fft_coefs[j].offset >> four_i;
|
|
1589 ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel;
|
|
1590
|
|
1591 if (offset < q->frequency_range) {
|
|
1592 if (offset < 2)
|
|
1593 tone.cutoff = offset;
|
|
1594 else
|
|
1595 tone.cutoff = (offset >= 60) ? 3 : 2;
|
|
1596
|
|
1597 tone.level = (q->fft_coefs[j].exp < 0) ? 0.0 : fft_tone_level_table[q->superblocktype_2_3 ? 0 : 1][q->fft_coefs[j].exp & 63];
|
|
1598 tone.samples_im = &q->fft.samples_im[ch][offset];
|
|
1599 tone.samples_re = &q->fft.samples_re[ch][offset];
|
|
1600 tone.table = (float*)fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
|
|
1601 tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
|
|
1602 tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
|
|
1603 tone.duration = i;
|
|
1604 tone.time_index = 0;
|
|
1605
|
|
1606 qdm2_fft_generate_tone(q, &tone);
|
|
1607 }
|
|
1608 }
|
|
1609 q->fft_coefs_min_index[i] = j;
|
|
1610 }
|
|
1611 }
|
|
1612
|
|
1613
|
|
1614 static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet)
|
|
1615 {
|
|
1616 const int n = 1 << (q->fft_order - 1);
|
|
1617 const int n2 = n >> 1;
|
|
1618 const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.25f : 0.50f;
|
|
1619 float c, s, f0, f1, f2, f3;
|
|
1620 int i, j;
|
|
1621
|
|
1622 /* pre rotation (or something like that) */
|
|
1623 for (i=1; i < n2; i++) {
|
|
1624 j = (n - i);
|
|
1625 c = q->exptab[i].re;
|
|
1626 s = -q->exptab[i].im;
|
|
1627 f0 = (q->fft.samples_re[channel][i] - q->fft.samples_re[channel][j]) * gain;
|
|
1628 f1 = (q->fft.samples_im[channel][i] + q->fft.samples_im[channel][j]) * gain;
|
|
1629 f2 = (q->fft.samples_re[channel][i] + q->fft.samples_re[channel][j]) * gain;
|
|
1630 f3 = (q->fft.samples_im[channel][i] - q->fft.samples_im[channel][j]) * gain;
|
|
1631 q->fft.complex[i].re = s * f0 - c * f1 + f2;
|
|
1632 q->fft.complex[i].im = c * f0 + s * f1 + f3;
|
|
1633 q->fft.complex[j].re = -s * f0 + c * f1 + f2;
|
|
1634 q->fft.complex[j].im = c * f0 + s * f1 - f3;
|
|
1635 }
|
|
1636
|
|
1637 q->fft.complex[ 0].re = q->fft.samples_re[channel][ 0] * gain * 2.0;
|
|
1638 q->fft.complex[ 0].im = q->fft.samples_re[channel][ 0] * gain * 2.0;
|
|
1639 q->fft.complex[n2].re = q->fft.samples_re[channel][n2] * gain * 2.0;
|
|
1640 q->fft.complex[n2].im = -q->fft.samples_im[channel][n2] * gain * 2.0;
|
|
1641
|
|
1642 ff_fft_permute(&q->fft_ctx, (FFTComplex *) q->fft.complex);
|
|
1643 ff_fft_calc (&q->fft_ctx, (FFTComplex *) q->fft.complex);
|
|
1644 /* add samples to output buffer */
|
|
1645 for (i = 0; i < ((q->fft_frame_size + 15) & ~15); i++)
|
|
1646 q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex)[i];
|
|
1647 }
|
|
1648
|
|
1649
|
|
1650 /**
|
|
1651 * @param q context
|
|
1652 * @param index subpacket number
|
|
1653 */
|
|
1654 static void qdm2_synthesis_filter (QDM2Context *q, int index)
|
|
1655 {
|
|
1656 OUT_INT samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
|
|
1657 int i, k, ch, sb_used, sub_sampling, dither_state = 0;
|
|
1658
|
|
1659 /* copy sb_samples */
|
|
1660 sb_used = QDM2_SB_USED(q->sub_sampling);
|
|
1661
|
|
1662 for (ch = 0; ch < q->channels; ch++)
|
|
1663 for (i = 0; i < 8; i++)
|
|
1664 for (k=sb_used; k < SBLIMIT; k++)
|
|
1665 q->sb_samples[ch][(8 * index) + i][k] = 0;
|
|
1666
|
|
1667 for (ch = 0; ch < q->nb_channels; ch++) {
|
|
1668 OUT_INT *samples_ptr = samples + ch;
|
|
1669
|
|
1670 for (i = 0; i < 8; i++) {
|
|
1671 ff_mpa_synth_filter(q->synth_buf[ch], &(q->synth_buf_offset[ch]),
|
|
1672 mpa_window, &dither_state,
|
|
1673 samples_ptr, q->nb_channels,
|
|
1674 q->sb_samples[ch][(8 * index) + i]);
|
|
1675 samples_ptr += 32 * q->nb_channels;
|
|
1676 }
|
|
1677 }
|
|
1678
|
|
1679 /* add samples to output buffer */
|
|
1680 sub_sampling = (4 >> q->sub_sampling);
|
|
1681
|
|
1682 for (ch = 0; ch < q->channels; ch++)
|
|
1683 for (i = 0; i < q->frame_size; i++)
|
|
1684 q->output_buffer[q->channels * i + ch] += (float)(samples[q->nb_channels * sub_sampling * i + ch] >> (sizeof(OUT_INT)*8-16));
|
|
1685 }
|
|
1686
|
|
1687
|
|
1688 /**
|
|
1689 * Init static data (does not depend on specific file)
|
|
1690 *
|
|
1691 * @param q context
|
|
1692 */
|
|
1693 void qdm2_init(QDM2Context *q) {
|
|
1694 static int inited = 0;
|
|
1695
|
|
1696 if (inited != 0)
|
|
1697 return;
|
|
1698 inited = 1;
|
|
1699
|
|
1700 qdm2_init_vlc();
|
|
1701 ff_mpa_synth_init(mpa_window);
|
|
1702 softclip_table_init();
|
|
1703 rnd_table_init();
|
|
1704 init_noise_samples();
|
|
1705
|
|
1706 av_log(NULL, AV_LOG_DEBUG, "init done\n");
|
|
1707 }
|
|
1708
|
|
1709
|
|
1710 #if 0
|
|
1711 static void dump_context(QDM2Context *q)
|
|
1712 {
|
|
1713 int i;
|
|
1714 #define PRINT(a,b) av_log(NULL,AV_LOG_DEBUG," %s = %d\n", a, b);
|
|
1715 PRINT("compressed_data",q->compressed_data);
|
|
1716 PRINT("compressed_size",q->compressed_size);
|
|
1717 PRINT("frame_size",q->frame_size);
|
|
1718 PRINT("checksum_size",q->checksum_size);
|
|
1719 PRINT("channels",q->channels);
|
|
1720 PRINT("nb_channels",q->nb_channels);
|
|
1721 PRINT("fft_frame_size",q->fft_frame_size);
|
|
1722 PRINT("fft_size",q->fft_size);
|
|
1723 PRINT("sub_sampling",q->sub_sampling);
|
|
1724 PRINT("fft_order",q->fft_order);
|
|
1725 PRINT("group_order",q->group_order);
|
|
1726 PRINT("group_size",q->group_size);
|
|
1727 PRINT("sub_packet",q->sub_packet);
|
|
1728 PRINT("frequency_range",q->frequency_range);
|
|
1729 PRINT("has_errors",q->has_errors);
|
|
1730 PRINT("fft_tone_end",q->fft_tone_end);
|
|
1731 PRINT("fft_tone_start",q->fft_tone_start);
|
|
1732 PRINT("fft_coefs_index",q->fft_coefs_index);
|
|
1733 PRINT("coeff_per_sb_select",q->coeff_per_sb_select);
|
|
1734 PRINT("cm_table_select",q->cm_table_select);
|
|
1735 PRINT("noise_idx",q->noise_idx);
|
|
1736
|
|
1737 for (i = q->fft_tone_start; i < q->fft_tone_end; i++)
|
|
1738 {
|
|
1739 FFTTone *t = &q->fft_tones[i];
|
2967
|
1740
|
2914
|
1741 av_log(NULL,AV_LOG_DEBUG,"Tone (%d) dump:\n", i);
|
|
1742 av_log(NULL,AV_LOG_DEBUG," level = %f\n", t->level);
|
|
1743 // PRINT(" level", t->level);
|
|
1744 PRINT(" phase", t->phase);
|
|
1745 PRINT(" phase_shift", t->phase_shift);
|
|
1746 PRINT(" duration", t->duration);
|
|
1747 PRINT(" samples_im", t->samples_im);
|
|
1748 PRINT(" samples_re", t->samples_re);
|
|
1749 PRINT(" table", t->table);
|
|
1750 }
|
|
1751
|
|
1752 }
|
|
1753 #endif
|
|
1754
|
|
1755
|
|
1756 /**
|
|
1757 * Init parameters from codec extradata
|
|
1758 */
|
|
1759 static int qdm2_decode_init(AVCodecContext *avctx)
|
|
1760 {
|
|
1761 QDM2Context *s = avctx->priv_data;
|
|
1762 uint8_t *extradata;
|
|
1763 int extradata_size;
|
|
1764 int tmp_val, tmp, size;
|
|
1765 int i;
|
|
1766 float alpha;
|
2967
|
1767
|
2914
|
1768 /* extradata parsing
|
2967
|
1769
|
2914
|
1770 Structure:
|
|
1771 wave {
|
|
1772 frma (QDM2)
|
|
1773 QDCA
|
|
1774 QDCP
|
|
1775 }
|
2967
|
1776
|
2914
|
1777 32 size (including this field)
|
|
1778 32 tag (=frma)
|
|
1779 32 type (=QDM2 or QDMC)
|
2967
|
1780
|
2914
|
1781 32 size (including this field, in bytes)
|
|
1782 32 tag (=QDCA) // maybe mandatory parameters
|
|
1783 32 unknown (=1)
|
|
1784 32 channels (=2)
|
|
1785 32 samplerate (=44100)
|
|
1786 32 bitrate (=96000)
|
|
1787 32 block size (=4096)
|
|
1788 32 frame size (=256) (for one channel)
|
|
1789 32 packet size (=1300)
|
2967
|
1790
|
2914
|
1791 32 size (including this field, in bytes)
|
|
1792 32 tag (=QDCP) // maybe some tuneable parameters
|
|
1793 32 float1 (=1.0)
|
|
1794 32 zero ?
|
|
1795 32 float2 (=1.0)
|
|
1796 32 float3 (=1.0)
|
|
1797 32 unknown (27)
|
|
1798 32 unknown (8)
|
|
1799 32 zero ?
|
|
1800 */
|
|
1801
|
|
1802 if (!avctx->extradata || (avctx->extradata_size < 48)) {
|
|
1803 av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
|
|
1804 return -1;
|
|
1805 }
|
|
1806
|
|
1807 extradata = avctx->extradata;
|
|
1808 extradata_size = avctx->extradata_size;
|
|
1809
|
|
1810 while (extradata_size > 7) {
|
|
1811 if (!memcmp(extradata, "frmaQDM", 7))
|
|
1812 break;
|
|
1813 extradata++;
|
|
1814 extradata_size--;
|
|
1815 }
|
|
1816
|
|
1817 if (extradata_size < 12) {
|
|
1818 av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
|
|
1819 extradata_size);
|
|
1820 return -1;
|
|
1821 }
|
|
1822
|
|
1823 if (memcmp(extradata, "frmaQDM", 7)) {
|
|
1824 av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n");
|
|
1825 return -1;
|
|
1826 }
|
|
1827
|
|
1828 if (extradata[7] == 'C') {
|
|
1829 // s->is_qdmc = 1;
|
|
1830 av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n");
|
|
1831 return -1;
|
|
1832 }
|
|
1833
|
|
1834 extradata += 8;
|
|
1835 extradata_size -= 8;
|
|
1836
|
|
1837 size = BE_32(extradata);
|
|
1838
|
|
1839 if(size > extradata_size){
|
|
1840 av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
|
|
1841 extradata_size, size);
|
|
1842 return -1;
|
|
1843 }
|
|
1844
|
|
1845 extradata += 4;
|
|
1846 av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
|
|
1847 if (BE_32(extradata) != MKBETAG('Q','D','C','A')) {
|
|
1848 av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
|
|
1849 return -1;
|
|
1850 }
|
|
1851
|
|
1852 extradata += 8;
|
|
1853
|
|
1854 avctx->channels = s->nb_channels = s->channels = BE_32(extradata);
|
|
1855 extradata += 4;
|
|
1856
|
|
1857 avctx->sample_rate = BE_32(extradata);
|
|
1858 extradata += 4;
|
|
1859
|
|
1860 avctx->bit_rate = BE_32(extradata);
|
|
1861 extradata += 4;
|
|
1862
|
|
1863 s->group_size = BE_32(extradata);
|
|
1864 extradata += 4;
|
|
1865
|
|
1866 s->fft_size = BE_32(extradata);
|
|
1867 extradata += 4;
|
|
1868
|
|
1869 s->checksum_size = BE_32(extradata);
|
|
1870 extradata += 4;
|
|
1871
|
|
1872 s->fft_order = av_log2(s->fft_size) + 1;
|
|
1873 s->fft_frame_size = 2 * s->fft_size; // complex has two floats
|
|
1874
|
|
1875 // something like max decodable tones
|
|
1876 s->group_order = av_log2(s->group_size) + 1;
|
|
1877 s->frame_size = s->group_size / 16; // 16 iterations per super block
|
|
1878
|
2954
|
1879 s->sub_sampling = s->fft_order - 7;
|
2914
|
1880 s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
|
2967
|
1881
|
2914
|
1882 switch ((s->sub_sampling * 2 + s->channels - 1)) {
|
|
1883 case 0: tmp = 40; break;
|
|
1884 case 1: tmp = 48; break;
|
|
1885 case 2: tmp = 56; break;
|
|
1886 case 3: tmp = 72; break;
|
|
1887 case 4: tmp = 80; break;
|
|
1888 case 5: tmp = 100;break;
|
|
1889 default: tmp=s->sub_sampling; break;
|
|
1890 }
|
|
1891 tmp_val = 0;
|
|
1892 if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1;
|
|
1893 if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2;
|
|
1894 if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3;
|
|
1895 if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4;
|
|
1896 s->cm_table_select = tmp_val;
|
|
1897
|
|
1898 if (s->sub_sampling == 0)
|
2954
|
1899 tmp = 7999;
|
2914
|
1900 else
|
|
1901 tmp = ((-(s->sub_sampling -1)) & 8000) + 20000;
|
|
1902 /*
|
2954
|
1903 0: 7999 -> 0
|
2914
|
1904 1: 20000 -> 2
|
|
1905 2: 28000 -> 2
|
|
1906 */
|
|
1907 if (tmp < 8000)
|
|
1908 s->coeff_per_sb_select = 0;
|
|
1909 else if (tmp <= 16000)
|
|
1910 s->coeff_per_sb_select = 1;
|
|
1911 else
|
|
1912 s->coeff_per_sb_select = 2;
|
|
1913
|
2954
|
1914 // Fail on unknown fft order, if it's > 9 it can overflow s->exptab[]
|
|
1915 if ((s->fft_order < 7) || (s->fft_order > 9)) {
|
2914
|
1916 av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order);
|
2954
|
1917 return -1;
|
|
1918 }
|
2914
|
1919
|
|
1920 ff_fft_init(&s->fft_ctx, s->fft_order - 1, 1);
|
|
1921
|
|
1922 for (i = 1; i < (1 << (s->fft_order - 2)); i++) {
|
|
1923 alpha = 2 * M_PI * (float)i / (float)(1 << (s->fft_order - 1));
|
|
1924 s->exptab[i].re = cos(alpha);
|
|
1925 s->exptab[i].im = sin(alpha);
|
|
1926 }
|
|
1927
|
|
1928 qdm2_init(s);
|
2967
|
1929
|
2914
|
1930 // dump_context(s);
|
|
1931 return 0;
|
|
1932 }
|
|
1933
|
|
1934
|
|
1935 static int qdm2_decode_close(AVCodecContext *avctx)
|
|
1936 {
|
|
1937 QDM2Context *s = avctx->priv_data;
|
|
1938
|
|
1939 ff_fft_end(&s->fft_ctx);
|
2967
|
1940
|
2914
|
1941 return 0;
|
|
1942 }
|
|
1943
|
|
1944
|
|
1945 void qdm2_decode (QDM2Context *q, uint8_t *in, int16_t *out)
|
|
1946 {
|
|
1947 int ch, i;
|
|
1948 const int frame_size = (q->frame_size * q->channels);
|
2967
|
1949
|
2914
|
1950 /* select input buffer */
|
|
1951 q->compressed_data = in;
|
|
1952 q->compressed_size = q->checksum_size;
|
|
1953
|
|
1954 // dump_context(q);
|
|
1955
|
|
1956 /* copy old block, clear new block of output samples */
|
|
1957 memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
|
|
1958 memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
|
|
1959
|
|
1960 /* decode block of QDM2 compressed data */
|
|
1961 if (q->sub_packet == 0) {
|
|
1962 q->has_errors = 0; // zero it for a new super block
|
|
1963 av_log(NULL,AV_LOG_DEBUG,"Super block follows\n");
|
|
1964 qdm2_decode_super_block(q);
|
|
1965 }
|
|
1966
|
|
1967 /* parse sub packets */
|
|
1968 if (!q->has_errors) {
|
|
1969 if (q->sub_packet == 2)
|
|
1970 qdm2_decode_fft_packets(q);
|
|
1971
|
|
1972 qdm2_fft_tone_synthesizer(q, q->sub_packet);
|
|
1973 }
|
|
1974
|
|
1975 /* sound synthesis stage 1 (FFT) */
|
|
1976 for (ch = 0; ch < q->channels; ch++) {
|
|
1977 qdm2_calculate_fft(q, ch, q->sub_packet);
|
|
1978
|
|
1979 if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) {
|
|
1980 SAMPLES_NEEDED_2("has errors, and C list is not empty")
|
|
1981 return;
|
|
1982 }
|
|
1983 }
|
|
1984
|
|
1985 /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
|
|
1986 if (!q->has_errors && q->do_synth_filter)
|
|
1987 qdm2_synthesis_filter(q, q->sub_packet);
|
|
1988
|
|
1989 q->sub_packet = (q->sub_packet + 1) % 16;
|
|
1990
|
|
1991 /* clip and convert output float[] to 16bit signed samples */
|
|
1992 for (i = 0; i < frame_size; i++) {
|
|
1993 int value = (int)q->output_buffer[i];
|
|
1994
|
|
1995 if (value > SOFTCLIP_THRESHOLD)
|
|
1996 value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD];
|
|
1997 else if (value < -SOFTCLIP_THRESHOLD)
|
|
1998 value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD];
|
|
1999
|
|
2000 out[i] = value;
|
|
2001 }
|
|
2002 }
|
|
2003
|
|
2004
|
|
2005 static int qdm2_decode_frame(AVCodecContext *avctx,
|
|
2006 void *data, int *data_size,
|
|
2007 uint8_t *buf, int buf_size)
|
|
2008 {
|
|
2009 QDM2Context *s = avctx->priv_data;
|
|
2010
|
|
2011 if((buf == NULL) || (buf_size < s->checksum_size))
|
|
2012 return 0;
|
|
2013
|
|
2014 *data_size = s->channels * s->frame_size * sizeof(int16_t);
|
|
2015
|
|
2016 av_log(avctx, AV_LOG_DEBUG, "decode(%d): %p[%d] -> %p[%d]\n",
|
|
2017 buf_size, buf, s->checksum_size, data, *data_size);
|
|
2018
|
|
2019 qdm2_decode(s, buf, data);
|
|
2020
|
|
2021 // reading only when next superblock found
|
|
2022 if (s->sub_packet == 0) {
|
|
2023 return s->checksum_size;
|
|
2024 }
|
|
2025
|
|
2026 return 0;
|
|
2027 }
|
|
2028
|
|
2029 AVCodec qdm2_decoder =
|
|
2030 {
|
|
2031 .name = "qdm2",
|
|
2032 .type = CODEC_TYPE_AUDIO,
|
|
2033 .id = CODEC_ID_QDM2,
|
|
2034 .priv_data_size = sizeof(QDM2Context),
|
|
2035 .init = qdm2_decode_init,
|
|
2036 .close = qdm2_decode_close,
|
|
2037 .decode = qdm2_decode_frame,
|
|
2038 };
|