annotate atrac1.c @ 10669:aa4c90d19a59 libavcodec

10l, Fix YUV ljpeg that i just broke by mistake.
author michael
date Thu, 10 Dec 2009 19:41:47 +0000
parents ec0e313a00aa
children 3d011a01a6a0
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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 and probably SDDS 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 float* spectrum[2];
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61 DECLARE_ALIGNED_16(float, spec1[AT1_SU_SAMPLES]); ///< mdct buffer
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62 DECLARE_ALIGNED_16(float, spec2[AT1_SU_SAMPLES]); ///< mdct buffer
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63 DECLARE_ALIGNED_16(float, fst_qmf_delay[46]); ///< delay line for the 1st stacked QMF filter
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64 DECLARE_ALIGNED_16(float, snd_qmf_delay[46]); ///< delay line for the 2nd stacked QMF filter
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65 DECLARE_ALIGNED_16(float, last_qmf_delay[256+23]); ///< delay line for the last stacked QMF filter
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66 } AT1SUCtx;
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67
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68 /**
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69 * The atrac1 context, holds all needed parameters for decoding
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70 */
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71 typedef struct {
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72 AT1SUCtx SUs[AT1_MAX_CHANNELS]; ///< channel sound unit
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73 DECLARE_ALIGNED_16(float, spec[AT1_SU_SAMPLES]); ///< the mdct spectrum buffer
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74
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75 DECLARE_ALIGNED_16(float, low[256]);
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76 DECLARE_ALIGNED_16(float, mid[256]);
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77 DECLARE_ALIGNED_16(float, high[512]);
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78 float* bands[3];
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79 DECLARE_ALIGNED_16(float, out_samples[AT1_MAX_CHANNELS][AT1_SU_SAMPLES]);
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80 FFTContext mdct_ctx[3];
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81 int channels;
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82 DSPContext dsp;
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83 } AT1Ctx;
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84
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85 /** size of the transform in samples in the long mode for each QMF band */
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86 static const uint16_t samples_per_band[3] = {128, 128, 256};
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87 static const uint8_t mdct_long_nbits[3] = {7, 7, 8};
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88
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89
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90 static void at1_imdct(AT1Ctx *q, float *spec, float *out, int nbits,
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91 int rev_spec)
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92 {
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93 FFTContext* mdct_context = &q->mdct_ctx[nbits - 5 - (nbits > 6)];
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94 int transf_size = 1 << nbits;
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95
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96 if (rev_spec) {
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97 int i;
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98 for (i = 0; i < transf_size / 2; i++)
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99 FFSWAP(float, spec[i], spec[transf_size - 1 - i]);
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100 }
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101 ff_imdct_half(mdct_context, out, spec);
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102 }
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103
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104
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105 static int at1_imdct_block(AT1SUCtx* su, AT1Ctx *q)
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106 {
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107 int band_num, band_samples, log2_block_count, nbits, num_blocks, block_size;
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108 unsigned int start_pos, ref_pos = 0, pos = 0;
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109
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110 for (band_num = 0; band_num < AT1_QMF_BANDS; band_num++) {
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111 float *prev_buf;
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112 int j;
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113
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114 band_samples = samples_per_band[band_num];
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115 log2_block_count = su->log2_block_count[band_num];
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116
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117 /* number of mdct blocks in the current QMF band: 1 - for long mode */
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118 /* 4 for short mode(low/middle bands) and 8 for short mode(high band)*/
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119 num_blocks = 1 << log2_block_count;
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120
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121 if (num_blocks == 1) {
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122 /* mdct block size in samples: 128 (long mode, low & mid bands), */
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123 /* 256 (long mode, high band) and 32 (short mode, all bands) */
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124 block_size = band_samples >> log2_block_count;
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125
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126 /* calc transform size in bits according to the block_size_mode */
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127 nbits = mdct_long_nbits[band_num] - log2_block_count;
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128
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129 if (nbits != 5 && nbits != 7 && nbits != 8)
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130 return -1;
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131 } else {
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132 block_size = 32;
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133 nbits = 5;
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134 }
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135
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136 start_pos = 0;
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137 prev_buf = &su->spectrum[1][ref_pos + band_samples - 16];
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138 for (j=0; j < num_blocks; j++) {
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139 at1_imdct(q, &q->spec[pos], &su->spectrum[0][ref_pos + start_pos], nbits, band_num);
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140
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141 /* overlap and window */
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142 q->dsp.vector_fmul_window(&q->bands[band_num][start_pos], prev_buf,
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143 &su->spectrum[0][ref_pos + start_pos], ff_sine_32, 0, 16);
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144
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145 prev_buf = &su->spectrum[0][ref_pos+start_pos + 16];
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146 start_pos += block_size;
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147 pos += block_size;
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148 }
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149
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150 if (num_blocks == 1)
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151 memcpy(q->bands[band_num] + 32, &su->spectrum[0][ref_pos + 16], 240 * sizeof(float));
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152
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153 ref_pos += band_samples;
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154 }
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155
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156 /* Swap buffers so the mdct overlap works */
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157 FFSWAP(float*, su->spectrum[0], su->spectrum[1]);
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158
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159 return 0;
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160 }
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161
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162 /**
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163 * Parse the block size mode byte
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164 */
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165
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166 static int at1_parse_bsm(GetBitContext* gb, int log2_block_cnt[AT1_QMF_BANDS])
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167 {
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168 int log2_block_count_tmp, i;
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169
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170 for (i = 0; i < 2; i++) {
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171 /* low and mid band */
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172 log2_block_count_tmp = get_bits(gb, 2);
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173 if (log2_block_count_tmp & 1)
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174 return -1;
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175 log2_block_cnt[i] = 2 - log2_block_count_tmp;
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176 }
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177
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178 /* high band */
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179 log2_block_count_tmp = get_bits(gb, 2);
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180 if (log2_block_count_tmp != 0 && log2_block_count_tmp != 3)
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181 return -1;
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182 log2_block_cnt[IDX_HIGH_BAND] = 3 - log2_block_count_tmp;
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183
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184 skip_bits(gb, 2);
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185 return 0;
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186 }
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187
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188
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189 static int at1_unpack_dequant(GetBitContext* gb, AT1SUCtx* su,
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190 float spec[AT1_SU_SAMPLES])
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191 {
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192 int bits_used, band_num, bfu_num, i;
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193 uint8_t idwls[AT1_MAX_BFU]; ///< the word length indexes for each BFU
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194 uint8_t idsfs[AT1_MAX_BFU]; ///< the scalefactor indexes for each BFU
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195
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196 /* parse the info byte (2nd byte) telling how much BFUs were coded */
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197 su->num_bfus = bfu_amount_tab1[get_bits(gb, 3)];
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198
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199 /* calc number of consumed bits:
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200 num_BFUs * (idwl(4bits) + idsf(6bits)) + log2_block_count(8bits) + info_byte(8bits)
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201 + info_byte_copy(8bits) + log2_block_count_copy(8bits) */
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202 bits_used = su->num_bfus * 10 + 32 +
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203 bfu_amount_tab2[get_bits(gb, 2)] +
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204 (bfu_amount_tab3[get_bits(gb, 3)] << 1);
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205
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206 /* get word length index (idwl) for each BFU */
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207 for (i = 0; i < su->num_bfus; i++)
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208 idwls[i] = get_bits(gb, 4);
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209
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210 /* get scalefactor index (idsf) for each BFU */
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211 for (i = 0; i < su->num_bfus; i++)
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212 idsfs[i] = get_bits(gb, 6);
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213
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214 /* zero idwl/idsf for empty BFUs */
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215 for (i = su->num_bfus; i < AT1_MAX_BFU; i++)
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216 idwls[i] = idsfs[i] = 0;
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217
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218 /* read in the spectral data and reconstruct MDCT spectrum of this channel */
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219 for (band_num = 0; band_num < AT1_QMF_BANDS; band_num++) {
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220 for (bfu_num = bfu_bands_t[band_num]; bfu_num < bfu_bands_t[band_num+1]; bfu_num++) {
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221 int pos;
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222
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223 int num_specs = specs_per_bfu[bfu_num];
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224 int word_len = !!idwls[bfu_num] + idwls[bfu_num];
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225 float scale_factor = sf_table[idsfs[bfu_num]];
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226 bits_used += word_len * num_specs; /* add number of bits consumed by current BFU */
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227
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228 /* check for bitstream overflow */
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229 if (bits_used > AT1_SU_MAX_BITS)
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230 return -1;
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231
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232 /* get the position of the 1st spec according to the block size mode */
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233 pos = su->log2_block_count[band_num] ? bfu_start_short[bfu_num] : bfu_start_long[bfu_num];
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234
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235 if (word_len) {
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236 float max_quant = 1.0 / (float)((1 << (word_len - 1)) - 1);
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237
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238 for (i = 0; i < num_specs; i++) {
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239 /* read in a quantized spec and convert it to
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240 * signed int and then inverse quantization
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241 */
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242 spec[pos+i] = get_sbits(gb, word_len) * scale_factor * max_quant;
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243 }
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244 } else { /* word_len = 0 -> empty BFU, zero all specs in the emty BFU */
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245 memset(&spec[pos], 0, num_specs * sizeof(float));
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246 }
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247 }
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248 }
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249
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250 return 0;
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251 }
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252
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253
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254 void at1_subband_synthesis(AT1Ctx *q, AT1SUCtx* su, float *pOut)
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255 {
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256 float temp[256];
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257 float iqmf_temp[512 + 46];
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258
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259 /* combine low and middle bands */
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260 atrac_iqmf(q->bands[0], q->bands[1], 128, temp, su->fst_qmf_delay, iqmf_temp);
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261
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262 /* delay the signal of the high band by 23 samples */
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263 memcpy( su->last_qmf_delay, &su->last_qmf_delay[256], sizeof(float) * 23);
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264 memcpy(&su->last_qmf_delay[23], q->bands[2], sizeof(float) * 256);
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265
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266 /* combine (low + middle) and high bands */
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267 atrac_iqmf(temp, su->last_qmf_delay, 256, pOut, su->snd_qmf_delay, iqmf_temp);
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268 }
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269
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270
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271 static int atrac1_decode_frame(AVCodecContext *avctx, void *data,
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272 int *data_size, AVPacket *avpkt)
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273 {
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274 const uint8_t *buf = avpkt->data;
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275 int buf_size = avpkt->size;
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276 AT1Ctx *q = avctx->priv_data;
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277 int ch, ret, i;
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278 GetBitContext gb;
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279 float* samples = data;
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280
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281
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282 if (buf_size < 212 * q->channels) {
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283 av_log(q,AV_LOG_ERROR,"Not enought data to decode!\n");
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284 return -1;
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285 }
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286
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287 for (ch = 0; ch < q->channels; ch++) {
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288 AT1SUCtx* su = &q->SUs[ch];
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289
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290 init_get_bits(&gb, &buf[212 * ch], 212 * 8);
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291
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292 /* parse block_size_mode, 1st byte */
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293 ret = at1_parse_bsm(&gb, su->log2_block_count);
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294 if (ret < 0)
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295 return ret;
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296
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297 ret = at1_unpack_dequant(&gb, su, q->spec);
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298 if (ret < 0)
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299 return ret;
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300
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301 ret = at1_imdct_block(su, q);
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302 if (ret < 0)
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303 return ret;
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304 at1_subband_synthesis(q, su, q->out_samples[ch]);
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305 }
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306
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parents:
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307 /* round, convert to 16bit and interleave */
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308 if (q->channels == 1) {
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309 /* mono */
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310 q->dsp.vector_clipf(samples, q->out_samples[0], -32700.0 / (1 << 15),
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311 32700.0 / (1 << 15), AT1_SU_SAMPLES);
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312 } else {
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313 /* stereo */
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314 for (i = 0; i < AT1_SU_SAMPLES; i++) {
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315 samples[i * 2] = av_clipf(q->out_samples[0][i],
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diff changeset
316 -32700.0 / (1 << 15),
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317 32700.0 / (1 << 15));
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318 samples[i * 2 + 1] = av_clipf(q->out_samples[1][i],
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319 -32700.0 / (1 << 15),
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320 32700.0 / (1 << 15));
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321 }
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322 }
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323
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324 *data_size = q->channels * AT1_SU_SAMPLES * sizeof(*samples);
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325 return avctx->block_align;
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parents:
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326 }
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327
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328
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329 static av_cold int atrac1_decode_init(AVCodecContext *avctx)
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330 {
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331 AT1Ctx *q = avctx->priv_data;
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332
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333 avctx->sample_fmt = SAMPLE_FMT_FLT;
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334
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335 q->channels = avctx->channels;
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336
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337 /* Init the mdct transforms */
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338 ff_mdct_init(&q->mdct_ctx[0], 6, 1, -1.0/ (1 << 15));
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339 ff_mdct_init(&q->mdct_ctx[1], 8, 1, -1.0/ (1 << 15));
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340 ff_mdct_init(&q->mdct_ctx[2], 9, 1, -1.0/ (1 << 15));
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341
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342 ff_sine_window_init(ff_sine_32, 32);
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343
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344 atrac_generate_tables();
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345
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346 dsputil_init(&q->dsp, avctx);
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347
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348 q->bands[0] = q->low;
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349 q->bands[1] = q->mid;
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350 q->bands[2] = q->high;
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351
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352 /* Prepare the mdct overlap buffers */
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353 q->SUs[0].spectrum[0] = q->SUs[0].spec1;
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354 q->SUs[0].spectrum[1] = q->SUs[0].spec2;
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355 q->SUs[1].spectrum[0] = q->SUs[1].spec1;
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356 q->SUs[1].spectrum[1] = q->SUs[1].spec2;
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357
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358 return 0;
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359 }
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360
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361
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362 static av_cold int atrac1_decode_end(AVCodecContext * avctx) {
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363 AT1Ctx *q = avctx->priv_data;
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364
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365 ff_mdct_end(&q->mdct_ctx[0]);
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366 ff_mdct_end(&q->mdct_ctx[1]);
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367 ff_mdct_end(&q->mdct_ctx[2]);
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368 return 0;
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369 }
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370
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371
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372 AVCodec atrac1_decoder = {
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373 .name = "atrac1",
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374 .type = CODEC_TYPE_AUDIO,
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375 .id = CODEC_ID_ATRAC1,
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376 .priv_data_size = sizeof(AT1Ctx),
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377 .init = atrac1_decode_init,
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378 .close = atrac1_decode_end,
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379 .decode = atrac1_decode_frame,
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380 .long_name = NULL_IF_CONFIG_SMALL("Atrac 1 (Adaptive TRansform Acoustic Coding)"),
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381 };