0
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1 /*
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2 * The simplest mpeg audio layer 2 encoder
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3 * Copyright (c) 2000 Gerard Lantau.
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4 *
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5 * This program is free software; you can redistribute it and/or modify
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6 * it under the terms of the GNU General Public License as published by
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7 * the Free Software Foundation; either version 2 of the License, or
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8 * (at your option) any later version.
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9 *
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10 * This program is distributed in the hope that it will be useful,
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11 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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13 * GNU General Public License for more details.
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14 *
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15 * You should have received a copy of the GNU General Public License
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16 * along with this program; if not, write to the Free Software
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17 * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
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18 */
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64
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19 #include "avcodec.h"
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0
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20 #include <math.h>
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21 #include "mpegaudio.h"
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22
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84
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23 #define DCT_BITS 14 /* number of bits for the DCT */
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24 #define MUL(a,b) (((a) * (b)) >> DCT_BITS)
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25 #define FIX(a) ((int)((a) * (1 << DCT_BITS)))
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26
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27 #define SAMPLES_BUF_SIZE 4096
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28
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29 typedef struct MpegAudioContext {
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30 PutBitContext pb;
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31 int nb_channels;
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32 int freq, bit_rate;
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33 int lsf; /* 1 if mpeg2 low bitrate selected */
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34 int bitrate_index; /* bit rate */
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35 int freq_index;
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36 int frame_size; /* frame size, in bits, without padding */
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37 INT64 nb_samples; /* total number of samples encoded */
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38 /* padding computation */
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39 int frame_frac, frame_frac_incr, do_padding;
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40 short samples_buf[MPA_MAX_CHANNELS][SAMPLES_BUF_SIZE]; /* buffer for filter */
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41 int samples_offset[MPA_MAX_CHANNELS]; /* offset in samples_buf */
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42 int sb_samples[MPA_MAX_CHANNELS][3][12][SBLIMIT];
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43 unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3]; /* scale factors */
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44 /* code to group 3 scale factors */
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45 unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
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46 int sblimit; /* number of used subbands */
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47 const unsigned char *alloc_table;
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48 } MpegAudioContext;
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49
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0
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50 /* define it to use floats in quantization (I don't like floats !) */
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51 //#define USE_FLOATS
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52
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53 #include "mpegaudiotab.h"
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54
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55 int MPA_encode_init(AVCodecContext *avctx)
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56 {
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57 MpegAudioContext *s = avctx->priv_data;
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58 int freq = avctx->sample_rate;
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59 int bitrate = avctx->bit_rate;
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60 int channels = avctx->channels;
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84
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61 int i, v, table;
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0
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62 float a;
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63
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64 if (channels > 2)
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65 return -1;
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66 bitrate = bitrate / 1000;
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67 s->nb_channels = channels;
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68 s->freq = freq;
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69 s->bit_rate = bitrate * 1000;
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70 avctx->frame_size = MPA_FRAME_SIZE;
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71 avctx->key_frame = 1; /* always key frame */
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72
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73 /* encoding freq */
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74 s->lsf = 0;
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75 for(i=0;i<3;i++) {
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84
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76 if (mpa_freq_tab[i] == freq)
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0
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77 break;
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84
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78 if ((mpa_freq_tab[i] / 2) == freq) {
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0
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79 s->lsf = 1;
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80 break;
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81 }
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82 }
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83 if (i == 3)
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84 return -1;
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85 s->freq_index = i;
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86
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87 /* encoding bitrate & frequency */
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88 for(i=0;i<15;i++) {
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84
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89 if (mpa_bitrate_tab[s->lsf][1][i] == bitrate)
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0
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90 break;
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91 }
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92 if (i == 15)
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93 return -1;
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94 s->bitrate_index = i;
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95
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96 /* compute total header size & pad bit */
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97
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98 a = (float)(bitrate * 1000 * MPA_FRAME_SIZE) / (freq * 8.0);
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99 s->frame_size = ((int)a) * 8;
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100
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101 /* frame fractional size to compute padding */
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102 s->frame_frac = 0;
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103 s->frame_frac_incr = (int)((a - floor(a)) * 65536.0);
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104
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105 /* select the right allocation table */
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84
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106 table = l2_select_table(bitrate, s->nb_channels, freq, s->lsf);
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107
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0
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108 /* number of used subbands */
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109 s->sblimit = sblimit_table[table];
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110 s->alloc_table = alloc_tables[table];
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111
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112 #ifdef DEBUG
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113 printf("%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n",
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114 bitrate, freq, s->frame_size, table, s->frame_frac_incr);
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115 #endif
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116
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117 for(i=0;i<s->nb_channels;i++)
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118 s->samples_offset[i] = 0;
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119
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84
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120 for(i=0;i<257;i++) {
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121 int v;
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122 v = (mpa_enwindow[i] + 2) >> 2;
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123 filter_bank[i] = v;
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124 if ((i & 63) != 0)
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125 v = -v;
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126 if (i != 0)
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127 filter_bank[512 - i] = v;
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0
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128 }
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84
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129
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0
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130 for(i=0;i<64;i++) {
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131 v = (int)(pow(2.0, (3 - i) / 3.0) * (1 << 20));
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132 if (v <= 0)
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133 v = 1;
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134 scale_factor_table[i] = v;
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135 #ifdef USE_FLOATS
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136 scale_factor_inv_table[i] = pow(2.0, -(3 - i) / 3.0) / (float)(1 << 20);
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137 #else
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138 #define P 15
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139 scale_factor_shift[i] = 21 - P - (i / 3);
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140 scale_factor_mult[i] = (1 << P) * pow(2.0, (i % 3) / 3.0);
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141 #endif
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142 }
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143 for(i=0;i<128;i++) {
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144 v = i - 64;
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145 if (v <= -3)
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146 v = 0;
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147 else if (v < 0)
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148 v = 1;
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149 else if (v == 0)
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150 v = 2;
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151 else if (v < 3)
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152 v = 3;
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153 else
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154 v = 4;
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155 scale_diff_table[i] = v;
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156 }
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157
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158 for(i=0;i<17;i++) {
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159 v = quant_bits[i];
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160 if (v < 0)
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161 v = -v;
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162 else
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163 v = v * 3;
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164 total_quant_bits[i] = 12 * v;
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165 }
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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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84
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170 /* 32 point floating point IDCT without 1/sqrt(2) coef zero scaling */
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0
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171 static void idct32(int *out, int *tab, int sblimit, int left_shift)
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172 {
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173 int i, j;
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174 int *t, *t1, xr;
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175 const int *xp = costab32;
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176
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177 for(j=31;j>=3;j-=2) tab[j] += tab[j - 2];
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178
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179 t = tab + 30;
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180 t1 = tab + 2;
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181 do {
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182 t[0] += t[-4];
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183 t[1] += t[1 - 4];
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184 t -= 4;
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185 } while (t != t1);
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186
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187 t = tab + 28;
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188 t1 = tab + 4;
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189 do {
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190 t[0] += t[-8];
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191 t[1] += t[1-8];
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192 t[2] += t[2-8];
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193 t[3] += t[3-8];
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194 t -= 8;
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195 } while (t != t1);
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196
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197 t = tab;
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198 t1 = tab + 32;
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199 do {
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200 t[ 3] = -t[ 3];
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201 t[ 6] = -t[ 6];
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202
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203 t[11] = -t[11];
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204 t[12] = -t[12];
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205 t[13] = -t[13];
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206 t[15] = -t[15];
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207 t += 16;
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208 } while (t != t1);
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209
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210
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211 t = tab;
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212 t1 = tab + 8;
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213 do {
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214 int x1, x2, x3, x4;
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215
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216 x3 = MUL(t[16], FIX(SQRT2*0.5));
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217 x4 = t[0] - x3;
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218 x3 = t[0] + x3;
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219
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220 x2 = MUL(-(t[24] + t[8]), FIX(SQRT2*0.5));
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221 x1 = MUL((t[8] - x2), xp[0]);
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222 x2 = MUL((t[8] + x2), xp[1]);
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223
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224 t[ 0] = x3 + x1;
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225 t[ 8] = x4 - x2;
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226 t[16] = x4 + x2;
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227 t[24] = x3 - x1;
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228 t++;
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229 } while (t != t1);
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230
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231 xp += 2;
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232 t = tab;
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233 t1 = tab + 4;
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234 do {
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235 xr = MUL(t[28],xp[0]);
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236 t[28] = (t[0] - xr);
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237 t[0] = (t[0] + xr);
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238
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239 xr = MUL(t[4],xp[1]);
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240 t[ 4] = (t[24] - xr);
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241 t[24] = (t[24] + xr);
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242
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243 xr = MUL(t[20],xp[2]);
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244 t[20] = (t[8] - xr);
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245 t[ 8] = (t[8] + xr);
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246
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247 xr = MUL(t[12],xp[3]);
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248 t[12] = (t[16] - xr);
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249 t[16] = (t[16] + xr);
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250 t++;
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251 } while (t != t1);
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252 xp += 4;
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253
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254 for (i = 0; i < 4; i++) {
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255 xr = MUL(tab[30-i*4],xp[0]);
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256 tab[30-i*4] = (tab[i*4] - xr);
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257 tab[ i*4] = (tab[i*4] + xr);
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258
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259 xr = MUL(tab[ 2+i*4],xp[1]);
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260 tab[ 2+i*4] = (tab[28-i*4] - xr);
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261 tab[28-i*4] = (tab[28-i*4] + xr);
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262
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263 xr = MUL(tab[31-i*4],xp[0]);
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264 tab[31-i*4] = (tab[1+i*4] - xr);
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265 tab[ 1+i*4] = (tab[1+i*4] + xr);
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266
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267 xr = MUL(tab[ 3+i*4],xp[1]);
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268 tab[ 3+i*4] = (tab[29-i*4] - xr);
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269 tab[29-i*4] = (tab[29-i*4] + xr);
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270
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271 xp += 2;
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272 }
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273
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274 t = tab + 30;
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275 t1 = tab + 1;
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276 do {
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277 xr = MUL(t1[0], *xp);
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278 t1[0] = (t[0] - xr);
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279 t[0] = (t[0] + xr);
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280 t -= 2;
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281 t1 += 2;
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282 xp++;
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283 } while (t >= tab);
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284
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285 for(i=0;i<32;i++) {
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286 out[i] = tab[bitinv32[i]] << left_shift;
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287 }
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288 }
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289
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290 static void filter(MpegAudioContext *s, int ch, short *samples, int incr)
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291 {
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292 short *p, *q;
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293 int sum, offset, i, j, norm, n;
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294 short tmp[64];
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295 int tmp1[32];
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296 int *out;
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297
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298 // print_pow1(samples, 1152);
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299
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300 offset = s->samples_offset[ch];
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301 out = &s->sb_samples[ch][0][0][0];
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302 for(j=0;j<36;j++) {
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303 /* 32 samples at once */
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304 for(i=0;i<32;i++) {
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305 s->samples_buf[ch][offset + (31 - i)] = samples[0];
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306 samples += incr;
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307 }
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308
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309 /* filter */
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310 p = s->samples_buf[ch] + offset;
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311 q = filter_bank;
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312 /* maxsum = 23169 */
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313 for(i=0;i<64;i++) {
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314 sum = p[0*64] * q[0*64];
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315 sum += p[1*64] * q[1*64];
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316 sum += p[2*64] * q[2*64];
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317 sum += p[3*64] * q[3*64];
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318 sum += p[4*64] * q[4*64];
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319 sum += p[5*64] * q[5*64];
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320 sum += p[6*64] * q[6*64];
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321 sum += p[7*64] * q[7*64];
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322 tmp[i] = sum >> 14;
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323 p++;
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324 q++;
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325 }
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326 tmp1[0] = tmp[16];
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327 for( i=1; i<=16; i++ ) tmp1[i] = tmp[i+16]+tmp[16-i];
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328 for( i=17; i<=31; i++ ) tmp1[i] = tmp[i+16]-tmp[80-i];
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329
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330 /* integer IDCT 32 with normalization. XXX: There may be some
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331 overflow left */
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332 norm = 0;
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333 for(i=0;i<32;i++) {
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334 norm |= abs(tmp1[i]);
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335 }
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70
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336 n = av_log2(norm) - 12;
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0
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337 if (n > 0) {
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338 for(i=0;i<32;i++)
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339 tmp1[i] >>= n;
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340 } else {
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341 n = 0;
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342 }
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343
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344 idct32(out, tmp1, s->sblimit, n);
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345
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346 /* advance of 32 samples */
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347 offset -= 32;
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348 out += 32;
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349 /* handle the wrap around */
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350 if (offset < 0) {
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351 memmove(s->samples_buf[ch] + SAMPLES_BUF_SIZE - (512 - 32),
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352 s->samples_buf[ch], (512 - 32) * 2);
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353 offset = SAMPLES_BUF_SIZE - 512;
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354 }
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355 }
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356 s->samples_offset[ch] = offset;
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357
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358 // print_pow(s->sb_samples, 1152);
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359 }
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360
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361 static void compute_scale_factors(unsigned char scale_code[SBLIMIT],
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362 unsigned char scale_factors[SBLIMIT][3],
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363 int sb_samples[3][12][SBLIMIT],
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364 int sblimit)
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365 {
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366 int *p, vmax, v, n, i, j, k, code;
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367 int index, d1, d2;
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368 unsigned char *sf = &scale_factors[0][0];
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369
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370 for(j=0;j<sblimit;j++) {
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371 for(i=0;i<3;i++) {
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372 /* find the max absolute value */
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373 p = &sb_samples[i][0][j];
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374 vmax = abs(*p);
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375 for(k=1;k<12;k++) {
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376 p += SBLIMIT;
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377 v = abs(*p);
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378 if (v > vmax)
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379 vmax = v;
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380 }
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381 /* compute the scale factor index using log 2 computations */
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382 if (vmax > 0) {
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70
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383 n = av_log2(vmax);
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0
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384 /* n is the position of the MSB of vmax. now
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385 use at most 2 compares to find the index */
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386 index = (21 - n) * 3 - 3;
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387 if (index >= 0) {
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388 while (vmax <= scale_factor_table[index+1])
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389 index++;
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390 } else {
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391 index = 0; /* very unlikely case of overflow */
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392 }
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393 } else {
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394 index = 63;
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395 }
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396
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397 #if 0
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398 printf("%2d:%d in=%x %x %d\n",
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399 j, i, vmax, scale_factor_table[index], index);
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400 #endif
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401 /* store the scale factor */
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402 assert(index >=0 && index <= 63);
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403 sf[i] = index;
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404 }
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405
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406 /* compute the transmission factor : look if the scale factors
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407 are close enough to each other */
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408 d1 = scale_diff_table[sf[0] - sf[1] + 64];
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409 d2 = scale_diff_table[sf[1] - sf[2] + 64];
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410
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411 /* handle the 25 cases */
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412 switch(d1 * 5 + d2) {
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413 case 0*5+0:
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414 case 0*5+4:
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415 case 3*5+4:
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416 case 4*5+0:
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417 case 4*5+4:
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418 code = 0;
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419 break;
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420 case 0*5+1:
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421 case 0*5+2:
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422 case 4*5+1:
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423 case 4*5+2:
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424 code = 3;
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425 sf[2] = sf[1];
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426 break;
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427 case 0*5+3:
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428 case 4*5+3:
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429 code = 3;
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430 sf[1] = sf[2];
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431 break;
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432 case 1*5+0:
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433 case 1*5+4:
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434 case 2*5+4:
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435 code = 1;
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436 sf[1] = sf[0];
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437 break;
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438 case 1*5+1:
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439 case 1*5+2:
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440 case 2*5+0:
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441 case 2*5+1:
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442 case 2*5+2:
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443 code = 2;
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444 sf[1] = sf[2] = sf[0];
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445 break;
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446 case 2*5+3:
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447 case 3*5+3:
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448 code = 2;
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449 sf[0] = sf[1] = sf[2];
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450 break;
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451 case 3*5+0:
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452 case 3*5+1:
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453 case 3*5+2:
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454 code = 2;
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455 sf[0] = sf[2] = sf[1];
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456 break;
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457 case 1*5+3:
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458 code = 2;
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459 if (sf[0] > sf[2])
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460 sf[0] = sf[2];
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461 sf[1] = sf[2] = sf[0];
|
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462 break;
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463 default:
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464 abort();
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465 }
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466
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467 #if 0
|
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468 printf("%d: %2d %2d %2d %d %d -> %d\n", j,
|
|
469 sf[0], sf[1], sf[2], d1, d2, code);
|
|
470 #endif
|
|
471 scale_code[j] = code;
|
|
472 sf += 3;
|
|
473 }
|
|
474 }
|
|
475
|
|
476 /* The most important function : psycho acoustic module. In this
|
|
477 encoder there is basically none, so this is the worst you can do,
|
|
478 but also this is the simpler. */
|
|
479 static void psycho_acoustic_model(MpegAudioContext *s, short smr[SBLIMIT])
|
|
480 {
|
|
481 int i;
|
|
482
|
|
483 for(i=0;i<s->sblimit;i++) {
|
|
484 smr[i] = (int)(fixed_smr[i] * 10);
|
|
485 }
|
|
486 }
|
|
487
|
|
488
|
|
489 #define SB_NOTALLOCATED 0
|
|
490 #define SB_ALLOCATED 1
|
|
491 #define SB_NOMORE 2
|
|
492
|
|
493 /* Try to maximize the smr while using a number of bits inferior to
|
|
494 the frame size. I tried to make the code simpler, faster and
|
|
495 smaller than other encoders :-) */
|
|
496 static void compute_bit_allocation(MpegAudioContext *s,
|
|
497 short smr1[MPA_MAX_CHANNELS][SBLIMIT],
|
|
498 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
|
|
499 int *padding)
|
|
500 {
|
|
501 int i, ch, b, max_smr, max_ch, max_sb, current_frame_size, max_frame_size;
|
|
502 int incr;
|
|
503 short smr[MPA_MAX_CHANNELS][SBLIMIT];
|
|
504 unsigned char subband_status[MPA_MAX_CHANNELS][SBLIMIT];
|
|
505 const unsigned char *alloc;
|
|
506
|
|
507 memcpy(smr, smr1, s->nb_channels * sizeof(short) * SBLIMIT);
|
|
508 memset(subband_status, SB_NOTALLOCATED, s->nb_channels * SBLIMIT);
|
|
509 memset(bit_alloc, 0, s->nb_channels * SBLIMIT);
|
|
510
|
|
511 /* compute frame size and padding */
|
|
512 max_frame_size = s->frame_size;
|
|
513 s->frame_frac += s->frame_frac_incr;
|
|
514 if (s->frame_frac >= 65536) {
|
|
515 s->frame_frac -= 65536;
|
|
516 s->do_padding = 1;
|
|
517 max_frame_size += 8;
|
|
518 } else {
|
|
519 s->do_padding = 0;
|
|
520 }
|
|
521
|
|
522 /* compute the header + bit alloc size */
|
|
523 current_frame_size = 32;
|
|
524 alloc = s->alloc_table;
|
|
525 for(i=0;i<s->sblimit;i++) {
|
|
526 incr = alloc[0];
|
|
527 current_frame_size += incr * s->nb_channels;
|
|
528 alloc += 1 << incr;
|
|
529 }
|
|
530 for(;;) {
|
|
531 /* look for the subband with the largest signal to mask ratio */
|
|
532 max_sb = -1;
|
|
533 max_ch = -1;
|
|
534 max_smr = 0x80000000;
|
|
535 for(ch=0;ch<s->nb_channels;ch++) {
|
|
536 for(i=0;i<s->sblimit;i++) {
|
|
537 if (smr[ch][i] > max_smr && subband_status[ch][i] != SB_NOMORE) {
|
|
538 max_smr = smr[ch][i];
|
|
539 max_sb = i;
|
|
540 max_ch = ch;
|
|
541 }
|
|
542 }
|
|
543 }
|
|
544 #if 0
|
|
545 printf("current=%d max=%d max_sb=%d alloc=%d\n",
|
|
546 current_frame_size, max_frame_size, max_sb,
|
|
547 bit_alloc[max_sb]);
|
|
548 #endif
|
|
549 if (max_sb < 0)
|
|
550 break;
|
|
551
|
|
552 /* find alloc table entry (XXX: not optimal, should use
|
|
553 pointer table) */
|
|
554 alloc = s->alloc_table;
|
|
555 for(i=0;i<max_sb;i++) {
|
|
556 alloc += 1 << alloc[0];
|
|
557 }
|
|
558
|
|
559 if (subband_status[max_ch][max_sb] == SB_NOTALLOCATED) {
|
|
560 /* nothing was coded for this band: add the necessary bits */
|
|
561 incr = 2 + nb_scale_factors[s->scale_code[max_ch][max_sb]] * 6;
|
|
562 incr += total_quant_bits[alloc[1]];
|
|
563 } else {
|
|
564 /* increments bit allocation */
|
|
565 b = bit_alloc[max_ch][max_sb];
|
|
566 incr = total_quant_bits[alloc[b + 1]] -
|
|
567 total_quant_bits[alloc[b]];
|
|
568 }
|
|
569
|
|
570 if (current_frame_size + incr <= max_frame_size) {
|
|
571 /* can increase size */
|
|
572 b = ++bit_alloc[max_ch][max_sb];
|
|
573 current_frame_size += incr;
|
|
574 /* decrease smr by the resolution we added */
|
|
575 smr[max_ch][max_sb] = smr1[max_ch][max_sb] - quant_snr[alloc[b]];
|
|
576 /* max allocation size reached ? */
|
|
577 if (b == ((1 << alloc[0]) - 1))
|
|
578 subband_status[max_ch][max_sb] = SB_NOMORE;
|
|
579 else
|
|
580 subband_status[max_ch][max_sb] = SB_ALLOCATED;
|
|
581 } else {
|
|
582 /* cannot increase the size of this subband */
|
|
583 subband_status[max_ch][max_sb] = SB_NOMORE;
|
|
584 }
|
|
585 }
|
|
586 *padding = max_frame_size - current_frame_size;
|
|
587 assert(*padding >= 0);
|
|
588
|
|
589 #if 0
|
|
590 for(i=0;i<s->sblimit;i++) {
|
|
591 printf("%d ", bit_alloc[i]);
|
|
592 }
|
|
593 printf("\n");
|
|
594 #endif
|
|
595 }
|
|
596
|
|
597 /*
|
|
598 * Output the mpeg audio layer 2 frame. Note how the code is small
|
|
599 * compared to other encoders :-)
|
|
600 */
|
|
601 static void encode_frame(MpegAudioContext *s,
|
|
602 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
|
|
603 int padding)
|
|
604 {
|
|
605 int i, j, k, l, bit_alloc_bits, b, ch;
|
|
606 unsigned char *sf;
|
|
607 int q[3];
|
|
608 PutBitContext *p = &s->pb;
|
|
609
|
|
610 /* header */
|
|
611
|
|
612 put_bits(p, 12, 0xfff);
|
|
613 put_bits(p, 1, 1 - s->lsf); /* 1 = mpeg1 ID, 0 = mpeg2 lsf ID */
|
|
614 put_bits(p, 2, 4-2); /* layer 2 */
|
|
615 put_bits(p, 1, 1); /* no error protection */
|
|
616 put_bits(p, 4, s->bitrate_index);
|
|
617 put_bits(p, 2, s->freq_index);
|
|
618 put_bits(p, 1, s->do_padding); /* use padding */
|
|
619 put_bits(p, 1, 0); /* private_bit */
|
|
620 put_bits(p, 2, s->nb_channels == 2 ? MPA_STEREO : MPA_MONO);
|
|
621 put_bits(p, 2, 0); /* mode_ext */
|
|
622 put_bits(p, 1, 0); /* no copyright */
|
|
623 put_bits(p, 1, 1); /* original */
|
|
624 put_bits(p, 2, 0); /* no emphasis */
|
|
625
|
|
626 /* bit allocation */
|
|
627 j = 0;
|
|
628 for(i=0;i<s->sblimit;i++) {
|
|
629 bit_alloc_bits = s->alloc_table[j];
|
|
630 for(ch=0;ch<s->nb_channels;ch++) {
|
|
631 put_bits(p, bit_alloc_bits, bit_alloc[ch][i]);
|
|
632 }
|
|
633 j += 1 << bit_alloc_bits;
|
|
634 }
|
|
635
|
|
636 /* scale codes */
|
|
637 for(i=0;i<s->sblimit;i++) {
|
|
638 for(ch=0;ch<s->nb_channels;ch++) {
|
|
639 if (bit_alloc[ch][i])
|
|
640 put_bits(p, 2, s->scale_code[ch][i]);
|
|
641 }
|
|
642 }
|
|
643
|
|
644 /* scale factors */
|
|
645 for(i=0;i<s->sblimit;i++) {
|
|
646 for(ch=0;ch<s->nb_channels;ch++) {
|
|
647 if (bit_alloc[ch][i]) {
|
|
648 sf = &s->scale_factors[ch][i][0];
|
|
649 switch(s->scale_code[ch][i]) {
|
|
650 case 0:
|
|
651 put_bits(p, 6, sf[0]);
|
|
652 put_bits(p, 6, sf[1]);
|
|
653 put_bits(p, 6, sf[2]);
|
|
654 break;
|
|
655 case 3:
|
|
656 case 1:
|
|
657 put_bits(p, 6, sf[0]);
|
|
658 put_bits(p, 6, sf[2]);
|
|
659 break;
|
|
660 case 2:
|
|
661 put_bits(p, 6, sf[0]);
|
|
662 break;
|
|
663 }
|
|
664 }
|
|
665 }
|
|
666 }
|
|
667
|
|
668 /* quantization & write sub band samples */
|
|
669
|
|
670 for(k=0;k<3;k++) {
|
|
671 for(l=0;l<12;l+=3) {
|
|
672 j = 0;
|
|
673 for(i=0;i<s->sblimit;i++) {
|
|
674 bit_alloc_bits = s->alloc_table[j];
|
|
675 for(ch=0;ch<s->nb_channels;ch++) {
|
|
676 b = bit_alloc[ch][i];
|
|
677 if (b) {
|
|
678 int qindex, steps, m, sample, bits;
|
|
679 /* we encode 3 sub band samples of the same sub band at a time */
|
|
680 qindex = s->alloc_table[j+b];
|
|
681 steps = quant_steps[qindex];
|
|
682 for(m=0;m<3;m++) {
|
|
683 sample = s->sb_samples[ch][k][l + m][i];
|
|
684 /* divide by scale factor */
|
|
685 #ifdef USE_FLOATS
|
|
686 {
|
|
687 float a;
|
|
688 a = (float)sample * scale_factor_inv_table[s->scale_factors[ch][i][k]];
|
|
689 q[m] = (int)((a + 1.0) * steps * 0.5);
|
|
690 }
|
|
691 #else
|
|
692 {
|
|
693 int q1, e, shift, mult;
|
|
694 e = s->scale_factors[ch][i][k];
|
|
695 shift = scale_factor_shift[e];
|
|
696 mult = scale_factor_mult[e];
|
|
697
|
|
698 /* normalize to P bits */
|
|
699 if (shift < 0)
|
|
700 q1 = sample << (-shift);
|
|
701 else
|
|
702 q1 = sample >> shift;
|
|
703 q1 = (q1 * mult) >> P;
|
|
704 q[m] = ((q1 + (1 << P)) * steps) >> (P + 1);
|
|
705 }
|
|
706 #endif
|
|
707 if (q[m] >= steps)
|
|
708 q[m] = steps - 1;
|
|
709 assert(q[m] >= 0 && q[m] < steps);
|
|
710 }
|
|
711 bits = quant_bits[qindex];
|
|
712 if (bits < 0) {
|
|
713 /* group the 3 values to save bits */
|
|
714 put_bits(p, -bits,
|
|
715 q[0] + steps * (q[1] + steps * q[2]));
|
|
716 #if 0
|
|
717 printf("%d: gr1 %d\n",
|
|
718 i, q[0] + steps * (q[1] + steps * q[2]));
|
|
719 #endif
|
|
720 } else {
|
|
721 #if 0
|
|
722 printf("%d: gr3 %d %d %d\n",
|
|
723 i, q[0], q[1], q[2]);
|
|
724 #endif
|
|
725 put_bits(p, bits, q[0]);
|
|
726 put_bits(p, bits, q[1]);
|
|
727 put_bits(p, bits, q[2]);
|
|
728 }
|
|
729 }
|
|
730 }
|
|
731 /* next subband in alloc table */
|
|
732 j += 1 << bit_alloc_bits;
|
|
733 }
|
|
734 }
|
|
735 }
|
|
736
|
|
737 /* padding */
|
|
738 for(i=0;i<padding;i++)
|
|
739 put_bits(p, 1, 0);
|
|
740
|
|
741 /* flush */
|
|
742 flush_put_bits(p);
|
|
743 }
|
|
744
|
|
745 int MPA_encode_frame(AVCodecContext *avctx,
|
|
746 unsigned char *frame, int buf_size, void *data)
|
|
747 {
|
|
748 MpegAudioContext *s = avctx->priv_data;
|
|
749 short *samples = data;
|
|
750 short smr[MPA_MAX_CHANNELS][SBLIMIT];
|
|
751 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
|
|
752 int padding, i;
|
|
753
|
|
754 for(i=0;i<s->nb_channels;i++) {
|
|
755 filter(s, i, samples + i, s->nb_channels);
|
|
756 }
|
|
757
|
|
758 for(i=0;i<s->nb_channels;i++) {
|
|
759 compute_scale_factors(s->scale_code[i], s->scale_factors[i],
|
|
760 s->sb_samples[i], s->sblimit);
|
|
761 }
|
|
762 for(i=0;i<s->nb_channels;i++) {
|
|
763 psycho_acoustic_model(s, smr[i]);
|
|
764 }
|
|
765 compute_bit_allocation(s, smr, bit_alloc, &padding);
|
|
766
|
|
767 init_put_bits(&s->pb, frame, MPA_MAX_CODED_FRAME_SIZE, NULL, NULL);
|
|
768
|
|
769 encode_frame(s, bit_alloc, padding);
|
|
770
|
|
771 s->nb_samples += MPA_FRAME_SIZE;
|
|
772 return s->pb.buf_ptr - s->pb.buf;
|
|
773 }
|
|
774
|
|
775
|
|
776 AVCodec mp2_encoder = {
|
|
777 "mp2",
|
|
778 CODEC_TYPE_AUDIO,
|
|
779 CODEC_ID_MP2,
|
|
780 sizeof(MpegAudioContext),
|
|
781 MPA_encode_init,
|
|
782 MPA_encode_frame,
|
|
783 NULL,
|
|
784 };
|