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Rev | Author | Line No. | Line |
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4349 | Serge | 1 | /* |
2 | * ATRAC1 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 | |||
23 | /** |
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24 | * @file |
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25 | * ATRAC1 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 | |||
29 | /* Many thanks to Tim Craig for all the help! */ |
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30 | |||
31 | #include |
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32 | #include |
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33 | #include |
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34 | |||
35 | #include "libavutil/float_dsp.h" |
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36 | #include "avcodec.h" |
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37 | #include "get_bits.h" |
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38 | #include "fft.h" |
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39 | #include "internal.h" |
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40 | #include "sinewin.h" |
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41 | |||
42 | #include "atrac.h" |
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43 | #include "atrac1data.h" |
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44 | |||
45 | #define AT1_MAX_BFU 52 ///< max number of block floating units in a sound unit |
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46 | #define AT1_SU_SIZE 212 ///< number of bytes in a sound unit |
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47 | #define AT1_SU_SAMPLES 512 ///< number of samples in a sound unit |
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48 | #define AT1_FRAME_SIZE AT1_SU_SIZE * 2 |
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49 | #define AT1_SU_MAX_BITS AT1_SU_SIZE * 8 |
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50 | #define AT1_MAX_CHANNELS 2 |
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51 | |||
52 | #define AT1_QMF_BANDS 3 |
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53 | #define IDX_LOW_BAND 0 |
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54 | #define IDX_MID_BAND 1 |
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55 | #define IDX_HIGH_BAND 2 |
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56 | |||
57 | /** |
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58 | * Sound unit struct, one unit is used per channel |
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59 | */ |
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60 | typedef struct { |
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61 | int log2_block_count[AT1_QMF_BANDS]; ///< log2 number of blocks in a band |
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62 | int num_bfus; ///< number of Block Floating Units |
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63 | float* spectrum[2]; |
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64 | DECLARE_ALIGNED(32, float, spec1)[AT1_SU_SAMPLES]; ///< mdct buffer |
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65 | DECLARE_ALIGNED(32, float, spec2)[AT1_SU_SAMPLES]; ///< mdct buffer |
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66 | DECLARE_ALIGNED(32, float, fst_qmf_delay)[46]; ///< delay line for the 1st stacked QMF filter |
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67 | DECLARE_ALIGNED(32, float, snd_qmf_delay)[46]; ///< delay line for the 2nd stacked QMF filter |
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68 | DECLARE_ALIGNED(32, float, last_qmf_delay)[256+23]; ///< delay line for the last stacked QMF filter |
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69 | } AT1SUCtx; |
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70 | |||
71 | /** |
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72 | * The atrac1 context, holds all needed parameters for decoding |
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73 | */ |
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74 | typedef struct { |
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75 | AT1SUCtx SUs[AT1_MAX_CHANNELS]; ///< channel sound unit |
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76 | DECLARE_ALIGNED(32, float, spec)[AT1_SU_SAMPLES]; ///< the mdct spectrum buffer |
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77 | |||
78 | DECLARE_ALIGNED(32, float, low)[256]; |
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79 | DECLARE_ALIGNED(32, float, mid)[256]; |
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80 | DECLARE_ALIGNED(32, float, high)[512]; |
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81 | float* bands[3]; |
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82 | FFTContext mdct_ctx[3]; |
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83 | AVFloatDSPContext fdsp; |
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84 | } AT1Ctx; |
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85 | |||
86 | /** size of the transform in samples in the long mode for each QMF band */ |
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87 | static const uint16_t samples_per_band[3] = {128, 128, 256}; |
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88 | static const uint8_t mdct_long_nbits[3] = {7, 7, 8}; |
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89 | |||
90 | |||
91 | static void at1_imdct(AT1Ctx *q, float *spec, float *out, int nbits, |
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92 | int rev_spec) |
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93 | { |
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94 | FFTContext* mdct_context = &q->mdct_ctx[nbits - 5 - (nbits > 6)]; |
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95 | int transf_size = 1 << nbits; |
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96 | |||
97 | if (rev_spec) { |
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98 | int i; |
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99 | for (i = 0; i < transf_size / 2; i++) |
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100 | FFSWAP(float, spec[i], spec[transf_size - 1 - i]); |
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101 | } |
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102 | mdct_context->imdct_half(mdct_context, out, spec); |
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103 | } |
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104 | |||
105 | |||
106 | static int at1_imdct_block(AT1SUCtx* su, AT1Ctx *q) |
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107 | { |
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108 | int band_num, band_samples, log2_block_count, nbits, num_blocks, block_size; |
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109 | unsigned int start_pos, ref_pos = 0, pos = 0; |
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110 | |||
111 | for (band_num = 0; band_num < AT1_QMF_BANDS; band_num++) { |
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112 | float *prev_buf; |
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113 | int j; |
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114 | |||
115 | band_samples = samples_per_band[band_num]; |
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116 | log2_block_count = su->log2_block_count[band_num]; |
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117 | |||
118 | /* number of mdct blocks in the current QMF band: 1 - for long mode */ |
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119 | /* 4 for short mode(low/middle bands) and 8 for short mode(high band)*/ |
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120 | num_blocks = 1 << log2_block_count; |
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121 | |||
122 | if (num_blocks == 1) { |
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123 | /* mdct block size in samples: 128 (long mode, low & mid bands), */ |
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124 | /* 256 (long mode, high band) and 32 (short mode, all bands) */ |
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125 | block_size = band_samples >> log2_block_count; |
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126 | |||
127 | /* calc transform size in bits according to the block_size_mode */ |
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128 | nbits = mdct_long_nbits[band_num] - log2_block_count; |
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129 | |||
130 | if (nbits != 5 && nbits != 7 && nbits != 8) |
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131 | return AVERROR_INVALIDDATA; |
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132 | } else { |
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133 | block_size = 32; |
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134 | nbits = 5; |
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135 | } |
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136 | |||
137 | start_pos = 0; |
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138 | prev_buf = &su->spectrum[1][ref_pos + band_samples - 16]; |
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139 | for (j=0; j < num_blocks; j++) { |
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140 | at1_imdct(q, &q->spec[pos], &su->spectrum[0][ref_pos + start_pos], nbits, band_num); |
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141 | |||
142 | /* overlap and window */ |
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143 | q->fdsp.vector_fmul_window(&q->bands[band_num][start_pos], prev_buf, |
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144 | &su->spectrum[0][ref_pos + start_pos], ff_sine_32, 16); |
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145 | |||
146 | prev_buf = &su->spectrum[0][ref_pos+start_pos + 16]; |
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147 | start_pos += block_size; |
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148 | pos += block_size; |
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149 | } |
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150 | |||
151 | if (num_blocks == 1) |
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152 | memcpy(q->bands[band_num] + 32, &su->spectrum[0][ref_pos + 16], 240 * sizeof(float)); |
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153 | |||
154 | ref_pos += band_samples; |
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155 | } |
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156 | |||
157 | /* Swap buffers so the mdct overlap works */ |
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158 | FFSWAP(float*, su->spectrum[0], su->spectrum[1]); |
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159 | |||
160 | return 0; |
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161 | } |
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162 | |||
163 | /** |
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164 | * Parse the block size mode byte |
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165 | */ |
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166 | |||
167 | static int at1_parse_bsm(GetBitContext* gb, int log2_block_cnt[AT1_QMF_BANDS]) |
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168 | { |
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169 | int log2_block_count_tmp, i; |
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170 | |||
171 | for (i = 0; i < 2; i++) { |
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172 | /* low and mid band */ |
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173 | log2_block_count_tmp = get_bits(gb, 2); |
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174 | if (log2_block_count_tmp & 1) |
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175 | return AVERROR_INVALIDDATA; |
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176 | log2_block_cnt[i] = 2 - log2_block_count_tmp; |
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177 | } |
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178 | |||
179 | /* high band */ |
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180 | log2_block_count_tmp = get_bits(gb, 2); |
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181 | if (log2_block_count_tmp != 0 && log2_block_count_tmp != 3) |
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182 | return AVERROR_INVALIDDATA; |
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183 | log2_block_cnt[IDX_HIGH_BAND] = 3 - log2_block_count_tmp; |
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184 | |||
185 | skip_bits(gb, 2); |
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186 | return 0; |
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187 | } |
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188 | |||
189 | |||
190 | static int at1_unpack_dequant(GetBitContext* gb, AT1SUCtx* su, |
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191 | float spec[AT1_SU_SAMPLES]) |
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192 | { |
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193 | int bits_used, band_num, bfu_num, i; |
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194 | uint8_t idwls[AT1_MAX_BFU]; ///< the word length indexes for each BFU |
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195 | uint8_t idsfs[AT1_MAX_BFU]; ///< the scalefactor indexes for each BFU |
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196 | |||
197 | /* parse the info byte (2nd byte) telling how much BFUs were coded */ |
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198 | su->num_bfus = bfu_amount_tab1[get_bits(gb, 3)]; |
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199 | |||
200 | /* calc number of consumed bits: |
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201 | num_BFUs * (idwl(4bits) + idsf(6bits)) + log2_block_count(8bits) + info_byte(8bits) |
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202 | + info_byte_copy(8bits) + log2_block_count_copy(8bits) */ |
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203 | bits_used = su->num_bfus * 10 + 32 + |
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204 | bfu_amount_tab2[get_bits(gb, 2)] + |
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205 | (bfu_amount_tab3[get_bits(gb, 3)] << 1); |
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206 | |||
207 | /* get word length index (idwl) for each BFU */ |
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208 | for (i = 0; i < su->num_bfus; i++) |
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209 | idwls[i] = get_bits(gb, 4); |
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210 | |||
211 | /* get scalefactor index (idsf) for each BFU */ |
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212 | for (i = 0; i < su->num_bfus; i++) |
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213 | idsfs[i] = get_bits(gb, 6); |
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214 | |||
215 | /* zero idwl/idsf for empty BFUs */ |
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216 | for (i = su->num_bfus; i < AT1_MAX_BFU; i++) |
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217 | idwls[i] = idsfs[i] = 0; |
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218 | |||
219 | /* read in the spectral data and reconstruct MDCT spectrum of this channel */ |
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220 | for (band_num = 0; band_num < AT1_QMF_BANDS; band_num++) { |
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221 | for (bfu_num = bfu_bands_t[band_num]; bfu_num < bfu_bands_t[band_num+1]; bfu_num++) { |
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222 | int pos; |
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223 | |||
224 | int num_specs = specs_per_bfu[bfu_num]; |
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225 | int word_len = !!idwls[bfu_num] + idwls[bfu_num]; |
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226 | float scale_factor = ff_atrac_sf_table[idsfs[bfu_num]]; |
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227 | bits_used += word_len * num_specs; /* add number of bits consumed by current BFU */ |
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228 | |||
229 | /* check for bitstream overflow */ |
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230 | if (bits_used > AT1_SU_MAX_BITS) |
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231 | return AVERROR_INVALIDDATA; |
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232 | |||
233 | /* get the position of the 1st spec according to the block size mode */ |
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234 | pos = su->log2_block_count[band_num] ? bfu_start_short[bfu_num] : bfu_start_long[bfu_num]; |
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235 | |||
236 | if (word_len) { |
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237 | float max_quant = 1.0 / (float)((1 << (word_len - 1)) - 1); |
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238 | |||
239 | for (i = 0; i < num_specs; i++) { |
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240 | /* read in a quantized spec and convert it to |
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241 | * signed int and then inverse quantization |
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242 | */ |
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243 | spec[pos+i] = get_sbits(gb, word_len) * scale_factor * max_quant; |
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244 | } |
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245 | } else { /* word_len = 0 -> empty BFU, zero all specs in the empty BFU */ |
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246 | memset(&spec[pos], 0, num_specs * sizeof(float)); |
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247 | } |
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248 | } |
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249 | } |
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250 | |||
251 | return 0; |
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252 | } |
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253 | |||
254 | |||
255 | static void at1_subband_synthesis(AT1Ctx *q, AT1SUCtx* su, float *pOut) |
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256 | { |
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257 | float temp[256]; |
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258 | float iqmf_temp[512 + 46]; |
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259 | |||
260 | /* combine low and middle bands */ |
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261 | ff_atrac_iqmf(q->bands[0], q->bands[1], 128, temp, su->fst_qmf_delay, iqmf_temp); |
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262 | |||
263 | /* delay the signal of the high band by 23 samples */ |
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264 | memcpy( su->last_qmf_delay, &su->last_qmf_delay[256], sizeof(float) * 23); |
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265 | memcpy(&su->last_qmf_delay[23], q->bands[2], sizeof(float) * 256); |
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266 | |||
267 | /* combine (low + middle) and high bands */ |
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268 | ff_atrac_iqmf(temp, su->last_qmf_delay, 256, pOut, su->snd_qmf_delay, iqmf_temp); |
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269 | } |
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270 | |||
271 | |||
272 | static int atrac1_decode_frame(AVCodecContext *avctx, void *data, |
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273 | int *got_frame_ptr, AVPacket *avpkt) |
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274 | { |
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275 | AVFrame *frame = data; |
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276 | const uint8_t *buf = avpkt->data; |
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277 | int buf_size = avpkt->size; |
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278 | AT1Ctx *q = avctx->priv_data; |
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279 | int ch, ret; |
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280 | GetBitContext gb; |
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281 | |||
282 | |||
283 | if (buf_size < 212 * avctx->channels) { |
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284 | av_log(avctx, AV_LOG_ERROR, "Not enough data to decode!\n"); |
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285 | return AVERROR_INVALIDDATA; |
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286 | } |
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287 | |||
288 | /* get output buffer */ |
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289 | frame->nb_samples = AT1_SU_SAMPLES; |
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290 | if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) |
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291 | return ret; |
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292 | |||
293 | for (ch = 0; ch < avctx->channels; ch++) { |
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294 | AT1SUCtx* su = &q->SUs[ch]; |
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295 | |||
296 | init_get_bits(&gb, &buf[212 * ch], 212 * 8); |
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297 | |||
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 | |||
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 | |||
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, (float *)frame->extended_data[ch]); |
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311 | } |
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312 | |||
313 | *got_frame_ptr = 1; |
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314 | |||
315 | return avctx->block_align; |
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316 | } |
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317 | |||
318 | |||
319 | static av_cold int atrac1_decode_end(AVCodecContext * avctx) |
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320 | { |
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321 | AT1Ctx *q = avctx->priv_data; |
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322 | |||
323 | ff_mdct_end(&q->mdct_ctx[0]); |
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324 | ff_mdct_end(&q->mdct_ctx[1]); |
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325 | ff_mdct_end(&q->mdct_ctx[2]); |
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326 | |||
327 | return 0; |
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328 | } |
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329 | |||
330 | |||
331 | static av_cold int atrac1_decode_init(AVCodecContext *avctx) |
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332 | { |
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333 | AT1Ctx *q = avctx->priv_data; |
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334 | int ret; |
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335 | |||
336 | avctx->sample_fmt = AV_SAMPLE_FMT_FLTP; |
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337 | |||
338 | if (avctx->channels < 1 || avctx->channels > AT1_MAX_CHANNELS) { |
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339 | av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %d\n", |
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340 | avctx->channels); |
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341 | return AVERROR(EINVAL); |
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342 | } |
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343 | |||
344 | if (avctx->block_align <= 0) { |
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345 | av_log(avctx, AV_LOG_ERROR, "Unsupported block align."); |
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346 | return AVERROR_PATCHWELCOME; |
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347 | } |
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348 | |||
349 | /* Init the mdct transforms */ |
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350 | if ((ret = ff_mdct_init(&q->mdct_ctx[0], 6, 1, -1.0/ (1 << 15))) || |
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351 | (ret = ff_mdct_init(&q->mdct_ctx[1], 8, 1, -1.0/ (1 << 15))) || |
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352 | (ret = ff_mdct_init(&q->mdct_ctx[2], 9, 1, -1.0/ (1 << 15)))) { |
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353 | av_log(avctx, AV_LOG_ERROR, "Error initializing MDCT\n"); |
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354 | atrac1_decode_end(avctx); |
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355 | return ret; |
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356 | } |
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357 | |||
358 | ff_init_ff_sine_windows(5); |
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359 | |||
360 | ff_atrac_generate_tables(); |
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361 | |||
362 | avpriv_float_dsp_init(&q->fdsp, avctx->flags & CODEC_FLAG_BITEXACT); |
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363 | |||
364 | q->bands[0] = q->low; |
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365 | q->bands[1] = q->mid; |
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366 | q->bands[2] = q->high; |
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367 | |||
368 | /* Prepare the mdct overlap buffers */ |
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369 | q->SUs[0].spectrum[0] = q->SUs[0].spec1; |
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370 | q->SUs[0].spectrum[1] = q->SUs[0].spec2; |
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371 | q->SUs[1].spectrum[0] = q->SUs[1].spec1; |
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372 | q->SUs[1].spectrum[1] = q->SUs[1].spec2; |
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373 | |||
374 | return 0; |
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375 | } |
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376 | |||
377 | |||
378 | AVCodec ff_atrac1_decoder = { |
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379 | .name = "atrac1", |
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380 | .long_name = NULL_IF_CONFIG_SMALL("ATRAC1 (Adaptive TRansform Acoustic Coding)"), |
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381 | .type = AVMEDIA_TYPE_AUDIO, |
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382 | .id = AV_CODEC_ID_ATRAC1, |
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383 | .priv_data_size = sizeof(AT1Ctx), |
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384 | .init = atrac1_decode_init, |
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385 | .close = atrac1_decode_end, |
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386 | .decode = atrac1_decode_frame, |
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387 | .capabilities = CODEC_CAP_DR1, |
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388 | .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP, |
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389 | AV_SAMPLE_FMT_NONE }, |
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390 | };><>><>><>=>>>>>>>>>><>>>>>>><>>>>>><>>>><>>>>>>>>>>>>> |