10157
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1 /*
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2 * Atrac 1 compatible decoder
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3 * Copyright (c) 2009 Maxim Poliakovski
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4 * Copyright (c) 2009 Benjamin Larsson
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5 *
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6 * This file is part of FFmpeg.
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7 *
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8 * FFmpeg is free software; you can redistribute it and/or
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9 * modify it under the terms of the GNU Lesser General Public
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10 * License as published by the Free Software Foundation; either
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11 * version 2.1 of the License, or (at your option) any later version.
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12 *
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13 * FFmpeg is distributed in the hope that it will be useful,
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14 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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16 * Lesser General Public License for more details.
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17 *
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18 * You should have received a copy of the GNU Lesser General Public
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19 * License along with FFmpeg; if not, write to the Free Software
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20 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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21 */
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22
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23 /**
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24 * @file libavcodec/atrac1.c
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25 * Atrac 1 compatible decoder.
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26 * This decoder handles raw ATRAC1 data.
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27 */
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28
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29 /* Many thanks to Tim Craig for all the help! */
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30
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31 #include <math.h>
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32 #include <stddef.h>
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33 #include <stdio.h>
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34
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35 #include "avcodec.h"
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36 #include "get_bits.h"
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37 #include "dsputil.h"
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38
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39 #include "atrac.h"
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40 #include "atrac1data.h"
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41
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42 #define AT1_MAX_BFU 52 ///< max number of block floating units in a sound unit
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43 #define AT1_SU_SIZE 212 ///< number of bytes in a sound unit
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44 #define AT1_SU_SAMPLES 512 ///< number of samples in a sound unit
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45 #define AT1_FRAME_SIZE AT1_SU_SIZE * 2
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46 #define AT1_SU_MAX_BITS AT1_SU_SIZE * 8
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47 #define AT1_MAX_CHANNELS 2
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48
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49 #define AT1_QMF_BANDS 3
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50 #define IDX_LOW_BAND 0
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51 #define IDX_MID_BAND 1
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52 #define IDX_HIGH_BAND 2
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53
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54 /**
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55 * Sound unit struct, one unit is used per channel
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56 */
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57 typedef struct {
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58 int log2_block_count[AT1_QMF_BANDS]; ///< log2 number of blocks in a band
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59 int num_bfus; ///< number of Block Floating Units
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60 int idwls[AT1_MAX_BFU]; ///< the word length indexes for each BFU
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61 int idsfs[AT1_MAX_BFU]; ///< the scalefactor indexes for each BFU
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62 float* spectrum[2];
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63 DECLARE_ALIGNED_16(float,spec1[AT1_SU_SAMPLES]); ///< mdct buffer
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64 DECLARE_ALIGNED_16(float,spec2[AT1_SU_SAMPLES]); ///< mdct buffer
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65 DECLARE_ALIGNED_16(float,fst_qmf_delay[46]); ///< delay line for the 1st stacked QMF filter
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66 DECLARE_ALIGNED_16(float,snd_qmf_delay[46]); ///< delay line for the 2nd stacked QMF filter
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67 DECLARE_ALIGNED_16(float,last_qmf_delay[256+23]); ///< delay line for the last stacked QMF filter
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68 } AT1SUCtx;
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69
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70 /**
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71 * The atrac1 context, holds all needed parameters for decoding
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72 */
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73 typedef struct {
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74 AT1SUCtx SUs[AT1_MAX_CHANNELS]; ///< channel sound unit
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75 DECLARE_ALIGNED_16(float,spec[AT1_SU_SAMPLES]); ///< the mdct spectrum buffer
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76
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77 DECLARE_ALIGNED_16(float, low[256]);
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78 DECLARE_ALIGNED_16(float, mid[256]);
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79 DECLARE_ALIGNED_16(float,high[512]);
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80 float* bands[3];
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81 DECLARE_ALIGNED_16(float,out_samples[AT1_MAX_CHANNELS][AT1_SU_SAMPLES]);
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82 MDCTContext mdct_ctx[3];
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83 int channels;
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84 DSPContext dsp;
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85 } AT1Ctx;
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86
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87 DECLARE_ALIGNED_16(static float, short_window[32]);
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88
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89 /** size of the transform in samples in the long mode for each QMF band */
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90 static const uint16_t samples_per_band[3] = {128, 128, 256};
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91 static const uint8_t mdct_long_nbits[3] = {7, 7, 8};
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92
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93
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94 static void at1_imdct(AT1Ctx *q, float *spec, float *out, int nbits,
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95 int rev_spec)
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96 {
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97 MDCTContext* mdct_context;
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98 int transf_size = 1 << nbits;
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99
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100 mdct_context = &q->mdct_ctx[nbits - 5 - (nbits>6)];
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101
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102 if (rev_spec) {
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103 int i;
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104 for (i=0 ; i<transf_size/2 ; i++)
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105 FFSWAP(float, spec[i], spec[transf_size - 1 - i]);
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106 }
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107 ff_imdct_half(mdct_context, out, spec);
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108 }
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109
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110
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111 static int at1_imdct_block(AT1SUCtx* su, AT1Ctx *q)
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112 {
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113 int band_num, band_samples, log2_block_count, nbits, num_blocks, block_size;
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114 unsigned int start_pos, ref_pos=0, pos = 0;
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115
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116 for (band_num=0 ; band_num<AT1_QMF_BANDS ; band_num++) {
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117 band_samples = samples_per_band[band_num];
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118 log2_block_count = su->log2_block_count[band_num];
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119
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120 /* number of mdct blocks in the current QMF band: 1 - for long mode */
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121 /* 4 for short mode(low/middle bands) and 8 for short mode(high band)*/
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122 num_blocks = 1 << log2_block_count;
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123
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124 /* mdct block size in samples: 128 (long mode, low & mid bands), */
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125 /* 256 (long mode, high band) and 32 (short mode, all bands) */
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126 block_size = band_samples >> log2_block_count;
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127
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128 /* calc transform size in bits according to the block_size_mode */
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129 nbits = mdct_long_nbits[band_num] - log2_block_count;
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130
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131 if (nbits!=5 && nbits!=7 && nbits!=8)
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132 return -1;
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133
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134 if (num_blocks == 1) {
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135 /* long blocks */
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136 at1_imdct(q, &q->spec[pos], &su->spectrum[0][ref_pos], nbits, band_num);
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137 pos += block_size; // move to the next mdct block in the spectrum
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138
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139 /* overlap and window long blocks */
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140 q->dsp.vector_fmul_window(q->bands[band_num], &su->spectrum[1][ref_pos+band_samples-16],
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141 &su->spectrum[0][ref_pos], short_window, 0, 16);
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142 memcpy(q->bands[band_num]+32, &su->spectrum[0][ref_pos+16], 240 * sizeof(float));
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143
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144 } else {
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145 /* short blocks */
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146 float *prev_buf;
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147 start_pos = 0;
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148 prev_buf = &su->spectrum[1][ref_pos+band_samples-16];
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149 for (; num_blocks!=0 ; num_blocks--) {
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150 at1_imdct(q, &q->spec[pos], &su->spectrum[0][ref_pos+start_pos], 5, band_num);
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151
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152 /* overlap and window between short blocks */
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153 q->dsp.vector_fmul_window(&q->bands[band_num][start_pos], prev_buf,
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154 &su->spectrum[0][ref_pos+start_pos], short_window, 0, 16);
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155
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156 prev_buf = &su->spectrum[0][ref_pos+start_pos+16];
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157 start_pos += 32; // use hardcoded block_size
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158 pos += 32;
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159 }
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160 }
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161 ref_pos += band_samples;
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162 }
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163
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164 /* Swap buffers so the mdct overlap works */
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165 FFSWAP(float*, su->spectrum[0], su->spectrum[1]);
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166
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167 return 0;
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168 }
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169
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170 /**
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171 * Parse the block size mode byte
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172 */
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173
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174 static int at1_parse_bsm(GetBitContext* gb, int log2_block_cnt[AT1_QMF_BANDS])
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175 {
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176 int log2_block_count_tmp, i;
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177
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178 for(i=0 ; i<2 ; i++) {
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179 /* low and mid band */
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180 log2_block_count_tmp = get_bits(gb, 2);
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181 if (log2_block_count_tmp & 1)
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182 return -1;
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183 log2_block_cnt[i] = 2 - log2_block_count_tmp;
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184 }
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185
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186 /* high band */
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187 log2_block_count_tmp = get_bits(gb, 2);
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188 if (log2_block_count_tmp != 0 && log2_block_count_tmp != 3)
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189 return -1;
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190 log2_block_cnt[IDX_HIGH_BAND] = 3 - log2_block_count_tmp;
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191
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192 skip_bits(gb, 2);
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193 return 0;
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194 }
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195
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196
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197 static int at1_unpack_dequant(GetBitContext* gb, AT1SUCtx* su,
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198 float spec[AT1_SU_SAMPLES])
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199 {
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200 int bits_used, band_num, bfu_num, i;
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201
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202 /* parse the info byte (2nd byte) telling how much BFUs were coded */
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203 su->num_bfus = bfu_amount_tab1[get_bits(gb, 3)];
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204
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205 /* calc number of consumed bits:
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206 num_BFUs * (idwl(4bits) + idsf(6bits)) + log2_block_count(8bits) + info_byte(8bits)
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207 + info_byte_copy(8bits) + log2_block_count_copy(8bits) */
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208 bits_used = su->num_bfus * 10 + 32 +
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209 bfu_amount_tab2[get_bits(gb, 2)] +
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210 (bfu_amount_tab3[get_bits(gb, 3)] << 1);
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211
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212 /* get word length index (idwl) for each BFU */
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213 for (i=0 ; i<su->num_bfus ; i++)
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214 su->idwls[i] = get_bits(gb, 4);
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215
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216 /* get scalefactor index (idsf) for each BFU */
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217 for (i=0 ; i<su->num_bfus ; i++)
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218 su->idsfs[i] = get_bits(gb, 6);
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219
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220 /* zero idwl/idsf for empty BFUs */
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221 for (i = su->num_bfus; i < AT1_MAX_BFU; i++)
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222 su->idwls[i] = su->idsfs[i] = 0;
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223
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224 /* read in the spectral data and reconstruct MDCT spectrum of this channel */
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225 for (band_num=0 ; band_num<AT1_QMF_BANDS ; band_num++) {
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226 for (bfu_num=bfu_bands_t[band_num] ; bfu_num<bfu_bands_t[band_num+1] ; bfu_num++) {
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227 int pos;
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228
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229 int num_specs = specs_per_bfu[bfu_num];
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230 int word_len = !!su->idwls[bfu_num] + su->idwls[bfu_num];
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231 float scale_factor = sf_table[su->idsfs[bfu_num]];
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232 bits_used += word_len * num_specs; /* add number of bits consumed by current BFU */
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233
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234 /* check for bitstream overflow */
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235 if (bits_used > AT1_SU_MAX_BITS)
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236 return -1;
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237
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238 /* get the position of the 1st spec according to the block size mode */
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239 pos = su->log2_block_count[band_num] ? bfu_start_short[bfu_num] : bfu_start_long[bfu_num];
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240
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241 if (word_len) {
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242 float max_quant = 1.0 / (float)((1 << (word_len - 1)) - 1);
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243
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244 for (i=0 ; i<num_specs ; i++) {
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245 /* read in a quantized spec and convert it to
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246 * signed int and then inverse quantization
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247 */
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248 spec[pos+i] = get_sbits(gb, word_len) * scale_factor * max_quant;
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249 }
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250 } else { /* word_len = 0 -> empty BFU, zero all specs in the emty BFU */
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251 memset(&spec[pos], 0, num_specs*sizeof(float));
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252 }
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253 }
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254 }
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255
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256 return 0;
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257 }
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258
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259
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260 void at1_subband_synthesis(AT1Ctx *q, AT1SUCtx* su, float *pOut)
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261 {
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262 float temp[256];
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263 float iqmf_temp[512 + 46];
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264
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265 /* combine low and middle bands */
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266 atrac_iqmf(q->bands[0], q->bands[1], 128, temp, su->fst_qmf_delay, iqmf_temp);
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267
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268 /* delay the signal of the high band by 23 samples */
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269 memcpy( su->last_qmf_delay, &su->last_qmf_delay[256], sizeof(float)*23);
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270 memcpy(&su->last_qmf_delay[23], q->bands[2], sizeof(float)*256);
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271
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272 /* combine (low + middle) and high bands */
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273 atrac_iqmf(temp, su->last_qmf_delay, 256, pOut, su->snd_qmf_delay, iqmf_temp);
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274 }
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275
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276
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277 static int atrac1_decode_frame(AVCodecContext *avctx, void *data,
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278 int *data_size, AVPacket *avpkt)
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279 {
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280 const uint8_t *buf = avpkt->data;
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281 int buf_size = avpkt->size;
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282 AT1Ctx *q = avctx->priv_data;
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283 int ch, ret, i;
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284 GetBitContext gb;
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285 float* samples = data;
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286
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287
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288 if (buf_size < 212 * q->channels) {
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289 av_log(q,AV_LOG_ERROR,"Not enought data to decode!\n");
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290 return -1;
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291 }
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292
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293 for (ch=0 ; ch<q->channels ; ch++) {
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294 AT1SUCtx* su = &q->SUs[ch];
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295
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296 init_get_bits(&gb, &buf[212*ch], 212*8);
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297
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298 /* parse block_size_mode, 1st byte */
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299 ret = at1_parse_bsm(&gb, su->log2_block_count);
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300 if (ret < 0)
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301 return ret;
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302
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303 ret = at1_unpack_dequant(&gb, su, q->spec);
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304 if (ret < 0)
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305 return ret;
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306
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307 ret = at1_imdct_block(su, q);
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308 if (ret < 0)
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309 return ret;
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310 at1_subband_synthesis(q, su, q->out_samples[ch]);
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311 }
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312
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313 /* round, convert to 16bit and interleave */
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314 if (q->channels == 1) {
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315 /* mono */
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316 q->dsp.vector_clipf(samples, q->out_samples[0], -32700.0 / (1<<15),
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317 32700.0 / (1<<15), AT1_SU_SAMPLES);
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318 } else {
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319 /* stereo */
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320 for (i = 0; i < AT1_SU_SAMPLES; i++) {
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321 samples[i*2] = av_clipf(q->out_samples[0][i], -32700.0 / (1<<15),
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322 32700.0 / (1<<15));
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323 samples[i*2+1] = av_clipf(q->out_samples[1][i], -32700.0 / (1<<15),
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324 32700.0 / (1<<15));
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325 }
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326 }
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327
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328 *data_size = q->channels * AT1_SU_SAMPLES * sizeof(*samples);
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329 return avctx->block_align;
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330 }
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331
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332
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333 static av_cold int atrac1_decode_init(AVCodecContext *avctx)
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334 {
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335 AT1Ctx *q = avctx->priv_data;
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336
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337 avctx->sample_fmt = SAMPLE_FMT_FLT;
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338
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339 q->channels = avctx->channels;
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340
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341 /* Init the mdct transforms */
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342 ff_mdct_init(&q->mdct_ctx[0], 6, 1, -1.0/ (1<<15));
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343 ff_mdct_init(&q->mdct_ctx[1], 8, 1, -1.0/ (1<<15));
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344 ff_mdct_init(&q->mdct_ctx[2], 9, 1, -1.0/ (1<<15));
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345
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346 ff_sine_window_init(short_window, 32);
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347
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348 atrac_generate_tables();
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349
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350 dsputil_init(&q->dsp, avctx);
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351
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352 q->bands[0] = q->low;
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353 q->bands[1] = q->mid;
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354 q->bands[2] = q->high;
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355
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356 /* Prepare the mdct overlap buffers */
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357 q->SUs[0].spectrum[0] = q->SUs[0].spec1;
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358 q->SUs[0].spectrum[1] = q->SUs[0].spec2;
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359 q->SUs[1].spectrum[0] = q->SUs[1].spec1;
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360 q->SUs[1].spectrum[1] = q->SUs[1].spec2;
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361
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362 return 0;
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363 }
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364
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365 AVCodec atrac1_decoder = {
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366 .name = "atrac1",
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367 .type = CODEC_TYPE_AUDIO,
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368 .id = CODEC_ID_ATRAC1,
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369 .priv_data_size = sizeof(AT1Ctx),
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370 .init = atrac1_decode_init,
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371 .close = NULL,
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372 .decode = atrac1_decode_frame,
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373 .long_name = NULL_IF_CONFIG_SMALL("Atrac 1 (Adaptive TRansform Acoustic Coding)"),
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374 };
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