annotate alacenc.c @ 8488:f2590c8d5e9b libavcodec

And another sechole ...
author michael
date Fri, 26 Dec 2008 18:14:59 +0000
parents 5f48c58bfd51
children ac31a0265eb9
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1 /**
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2 * ALAC audio encoder
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3 * Copyright (c) 2008 Jaikrishnan Menon <realityman@gmx.net>
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4 *
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5 * This file is part of FFmpeg.
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6 *
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7 * FFmpeg is free software; you can redistribute it and/or
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8 * modify it under the terms of the GNU Lesser General Public
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9 * License as published by the Free Software Foundation; either
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10 * version 2.1 of the License, or (at your option) any later version.
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11 *
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12 * FFmpeg is distributed in the hope that it will be useful,
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13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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15 * Lesser General Public License for more details.
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16 *
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17 * You should have received a copy of the GNU Lesser General Public
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18 * License along with FFmpeg; if not, write to the Free Software
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19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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20 */
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21
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22 #include "avcodec.h"
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23 #include "bitstream.h"
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24 #include "dsputil.h"
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25 #include "lpc.h"
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26
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27 #define DEFAULT_FRAME_SIZE 4096
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28 #define DEFAULT_SAMPLE_SIZE 16
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29 #define MAX_CHANNELS 8
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30 #define ALAC_EXTRADATA_SIZE 36
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31 #define ALAC_FRAME_HEADER_SIZE 55
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32 #define ALAC_FRAME_FOOTER_SIZE 3
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33
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34 #define ALAC_ESCAPE_CODE 0x1FF
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35 #define ALAC_MAX_LPC_ORDER 30
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36 #define DEFAULT_MAX_PRED_ORDER 6
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37 #define DEFAULT_MIN_PRED_ORDER 4
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38 #define ALAC_MAX_LPC_PRECISION 9
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39 #define ALAC_MAX_LPC_SHIFT 9
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40
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41 #define ALAC_CHMODE_LEFT_RIGHT 0
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42 #define ALAC_CHMODE_LEFT_SIDE 1
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43 #define ALAC_CHMODE_RIGHT_SIDE 2
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44 #define ALAC_CHMODE_MID_SIDE 3
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45
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46 typedef struct RiceContext {
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47 int history_mult;
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48 int initial_history;
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49 int k_modifier;
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50 int rice_modifier;
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51 } RiceContext;
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52
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53 typedef struct LPCContext {
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54 int lpc_order;
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55 int lpc_coeff[ALAC_MAX_LPC_ORDER+1];
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56 int lpc_quant;
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57 } LPCContext;
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58
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59 typedef struct AlacEncodeContext {
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60 int compression_level;
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61 int min_prediction_order;
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62 int max_prediction_order;
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63 int max_coded_frame_size;
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64 int write_sample_size;
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65 int32_t sample_buf[MAX_CHANNELS][DEFAULT_FRAME_SIZE];
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66 int32_t predictor_buf[DEFAULT_FRAME_SIZE];
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67 int interlacing_shift;
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68 int interlacing_leftweight;
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69 PutBitContext pbctx;
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70 RiceContext rc;
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71 LPCContext lpc[MAX_CHANNELS];
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72 DSPContext dspctx;
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73 AVCodecContext *avctx;
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74 } AlacEncodeContext;
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75
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76
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77 static void init_sample_buffers(AlacEncodeContext *s, int16_t *input_samples)
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78 {
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79 int ch, i;
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80
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81 for(ch=0;ch<s->avctx->channels;ch++) {
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82 int16_t *sptr = input_samples + ch;
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83 for(i=0;i<s->avctx->frame_size;i++) {
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84 s->sample_buf[ch][i] = *sptr;
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85 sptr += s->avctx->channels;
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86 }
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87 }
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88 }
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89
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90 static void encode_scalar(AlacEncodeContext *s, int x, int k, int write_sample_size)
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91 {
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92 int divisor, q, r;
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93
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94 k = FFMIN(k, s->rc.k_modifier);
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95 divisor = (1<<k) - 1;
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96 q = x / divisor;
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97 r = x % divisor;
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98
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99 if(q > 8) {
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100 // write escape code and sample value directly
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101 put_bits(&s->pbctx, 9, ALAC_ESCAPE_CODE);
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102 put_bits(&s->pbctx, write_sample_size, x);
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103 } else {
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104 if(q)
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105 put_bits(&s->pbctx, q, (1<<q) - 1);
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106 put_bits(&s->pbctx, 1, 0);
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107
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108 if(k != 1) {
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109 if(r > 0)
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110 put_bits(&s->pbctx, k, r+1);
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111 else
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112 put_bits(&s->pbctx, k-1, 0);
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113 }
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114 }
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115 }
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116
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117 static void write_frame_header(AlacEncodeContext *s, int is_verbatim)
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118 {
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119 put_bits(&s->pbctx, 3, s->avctx->channels-1); // No. of channels -1
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120 put_bits(&s->pbctx, 16, 0); // Seems to be zero
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121 put_bits(&s->pbctx, 1, 1); // Sample count is in the header
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122 put_bits(&s->pbctx, 2, 0); // FIXME: Wasted bytes field
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123 put_bits(&s->pbctx, 1, is_verbatim); // Audio block is verbatim
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124 put_bits(&s->pbctx, 32, s->avctx->frame_size); // No. of samples in the frame
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125 }
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126
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127 static void calc_predictor_params(AlacEncodeContext *s, int ch)
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128 {
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129 int32_t coefs[MAX_LPC_ORDER][MAX_LPC_ORDER];
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130 int shift[MAX_LPC_ORDER];
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131 int opt_order;
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132
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133 opt_order = ff_lpc_calc_coefs(&s->dspctx, s->sample_buf[ch], s->avctx->frame_size, s->min_prediction_order, s->max_prediction_order,
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134 ALAC_MAX_LPC_PRECISION, coefs, shift, 1, ORDER_METHOD_EST, ALAC_MAX_LPC_SHIFT, 1);
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135
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136 s->lpc[ch].lpc_order = opt_order;
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137 s->lpc[ch].lpc_quant = shift[opt_order-1];
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138 memcpy(s->lpc[ch].lpc_coeff, coefs[opt_order-1], opt_order*sizeof(int));
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139 }
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140
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141 static int estimate_stereo_mode(int32_t *left_ch, int32_t *right_ch, int n)
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142 {
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143 int i, best;
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144 int32_t lt, rt;
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145 uint64_t sum[4];
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146 uint64_t score[4];
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147
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148 /* calculate sum of 2nd order residual for each channel */
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149 sum[0] = sum[1] = sum[2] = sum[3] = 0;
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150 for(i=2; i<n; i++) {
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151 lt = left_ch[i] - 2*left_ch[i-1] + left_ch[i-2];
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152 rt = right_ch[i] - 2*right_ch[i-1] + right_ch[i-2];
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153 sum[2] += FFABS((lt + rt) >> 1);
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154 sum[3] += FFABS(lt - rt);
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155 sum[0] += FFABS(lt);
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156 sum[1] += FFABS(rt);
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157 }
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158
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159 /* calculate score for each mode */
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160 score[0] = sum[0] + sum[1];
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161 score[1] = sum[0] + sum[3];
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162 score[2] = sum[1] + sum[3];
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163 score[3] = sum[2] + sum[3];
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164
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165 /* return mode with lowest score */
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166 best = 0;
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167 for(i=1; i<4; i++) {
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168 if(score[i] < score[best]) {
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169 best = i;
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170 }
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171 }
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172 return best;
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173 }
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174
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175 static void alac_stereo_decorrelation(AlacEncodeContext *s)
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176 {
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177 int32_t *left = s->sample_buf[0], *right = s->sample_buf[1];
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178 int i, mode, n = s->avctx->frame_size;
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179 int32_t tmp;
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180
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181 mode = estimate_stereo_mode(left, right, n);
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182
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183 switch(mode)
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184 {
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185 case ALAC_CHMODE_LEFT_RIGHT:
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186 s->interlacing_leftweight = 0;
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187 s->interlacing_shift = 0;
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188 break;
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189
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190 case ALAC_CHMODE_LEFT_SIDE:
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191 for(i=0; i<n; i++) {
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192 right[i] = left[i] - right[i];
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193 }
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194 s->interlacing_leftweight = 1;
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195 s->interlacing_shift = 0;
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196 break;
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197
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198 case ALAC_CHMODE_RIGHT_SIDE:
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199 for(i=0; i<n; i++) {
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200 tmp = right[i];
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201 right[i] = left[i] - right[i];
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202 left[i] = tmp + (right[i] >> 31);
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203 }
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204 s->interlacing_leftweight = 1;
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205 s->interlacing_shift = 31;
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206 break;
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207
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208 default:
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209 for(i=0; i<n; i++) {
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210 tmp = left[i];
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211 left[i] = (tmp + right[i]) >> 1;
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212 right[i] = tmp - right[i];
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213 }
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214 s->interlacing_leftweight = 1;
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215 s->interlacing_shift = 1;
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216 break;
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217 }
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218 }
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219
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220 static void alac_linear_predictor(AlacEncodeContext *s, int ch)
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221 {
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222 int i;
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223 LPCContext lpc = s->lpc[ch];
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224
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225 if(lpc.lpc_order == 31) {
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226 s->predictor_buf[0] = s->sample_buf[ch][0];
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227
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228 for(i=1; i<s->avctx->frame_size; i++)
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229 s->predictor_buf[i] = s->sample_buf[ch][i] - s->sample_buf[ch][i-1];
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230
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231 return;
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232 }
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233
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234 // generalised linear predictor
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235
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236 if(lpc.lpc_order > 0) {
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237 int32_t *samples = s->sample_buf[ch];
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238 int32_t *residual = s->predictor_buf;
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239
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240 // generate warm-up samples
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241 residual[0] = samples[0];
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242 for(i=1;i<=lpc.lpc_order;i++)
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243 residual[i] = samples[i] - samples[i-1];
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244
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245 // perform lpc on remaining samples
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246 for(i = lpc.lpc_order + 1; i < s->avctx->frame_size; i++) {
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247 int sum = 1 << (lpc.lpc_quant - 1), res_val, j;
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248
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249 for (j = 0; j < lpc.lpc_order; j++) {
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250 sum += (samples[lpc.lpc_order-j] - samples[0]) *
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251 lpc.lpc_coeff[j];
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252 }
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253
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254 sum >>= lpc.lpc_quant;
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255 sum += samples[0];
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256 residual[i] = (samples[lpc.lpc_order+1] - sum) << (32 - s->write_sample_size) >>
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257 (32 - s->write_sample_size);
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258 res_val = residual[i];
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259
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260 if(res_val) {
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261 int index = lpc.lpc_order - 1;
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262 int neg = (res_val < 0);
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263
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264 while(index >= 0 && (neg ? (res_val < 0):(res_val > 0))) {
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265 int val = samples[0] - samples[lpc.lpc_order - index];
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266 int sign = (val ? FFSIGN(val) : 0);
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267
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268 if(neg)
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269 sign*=-1;
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270
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271 lpc.lpc_coeff[index] -= sign;
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272 val *= sign;
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273 res_val -= ((val >> lpc.lpc_quant) *
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274 (lpc.lpc_order - index));
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275 index--;
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276 }
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277 }
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278 samples++;
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279 }
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280 }
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281 }
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282
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283 static void alac_entropy_coder(AlacEncodeContext *s)
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284 {
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285 unsigned int history = s->rc.initial_history;
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286 int sign_modifier = 0, i, k;
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287 int32_t *samples = s->predictor_buf;
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288
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289 for(i=0;i < s->avctx->frame_size;) {
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290 int x;
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291
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292 k = av_log2((history >> 9) + 3);
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293
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294 x = -2*(*samples)-1;
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295 x ^= (x>>31);
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296
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297 samples++;
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298 i++;
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299
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300 encode_scalar(s, x - sign_modifier, k, s->write_sample_size);
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301
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302 history += x * s->rc.history_mult
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303 - ((history * s->rc.history_mult) >> 9);
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304
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305 sign_modifier = 0;
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306 if(x > 0xFFFF)
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307 history = 0xFFFF;
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308
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309 if((history < 128) && (i < s->avctx->frame_size)) {
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310 unsigned int block_size = 0;
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311
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312 k = 7 - av_log2(history) + ((history + 16) >> 6);
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313
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314 while((*samples == 0) && (i < s->avctx->frame_size)) {
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315 samples++;
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316 i++;
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317 block_size++;
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318 }
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319 encode_scalar(s, block_size, k, 16);
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320
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321 sign_modifier = (block_size <= 0xFFFF);
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322
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323 history = 0;
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324 }
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325
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326 }
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327 }
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328
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329 static void write_compressed_frame(AlacEncodeContext *s)
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330 {
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331 int i, j;
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332
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333 /* only simple mid/side decorrelation supported as of now */
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334 if(s->avctx->channels == 2)
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335 alac_stereo_decorrelation(s);
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336 put_bits(&s->pbctx, 8, s->interlacing_shift);
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337 put_bits(&s->pbctx, 8, s->interlacing_leftweight);
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338
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339 for(i=0;i<s->avctx->channels;i++) {
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340
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341 calc_predictor_params(s, i);
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342
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343 put_bits(&s->pbctx, 4, 0); // prediction type : currently only type 0 has been RE'd
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344 put_bits(&s->pbctx, 4, s->lpc[i].lpc_quant);
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345
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346 put_bits(&s->pbctx, 3, s->rc.rice_modifier);
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347 put_bits(&s->pbctx, 5, s->lpc[i].lpc_order);
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348 // predictor coeff. table
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349 for(j=0;j<s->lpc[i].lpc_order;j++) {
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350 put_sbits(&s->pbctx, 16, s->lpc[i].lpc_coeff[j]);
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351 }
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352 }
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353
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354 // apply lpc and entropy coding to audio samples
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355
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356 for(i=0;i<s->avctx->channels;i++) {
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357 alac_linear_predictor(s, i);
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358 alac_entropy_coder(s);
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359 }
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360 }
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361
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362 static av_cold int alac_encode_init(AVCodecContext *avctx)
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363 {
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364 AlacEncodeContext *s = avctx->priv_data;
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365 uint8_t *alac_extradata = av_mallocz(ALAC_EXTRADATA_SIZE+1);
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366
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367 avctx->frame_size = DEFAULT_FRAME_SIZE;
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368 avctx->bits_per_coded_sample = DEFAULT_SAMPLE_SIZE;
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369
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370 if(avctx->sample_fmt != SAMPLE_FMT_S16) {
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371 av_log(avctx, AV_LOG_ERROR, "only pcm_s16 input samples are supported\n");
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372 return -1;
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373 }
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374
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375 // Set default compression level
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376 if(avctx->compression_level == FF_COMPRESSION_DEFAULT)
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377 s->compression_level = 1;
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378 else
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379 s->compression_level = av_clip(avctx->compression_level, 0, 1);
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380
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381 // Initialize default Rice parameters
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382 s->rc.history_mult = 40;
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383 s->rc.initial_history = 10;
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384 s->rc.k_modifier = 14;
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385 s->rc.rice_modifier = 4;
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386
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387 s->max_coded_frame_size = (ALAC_FRAME_HEADER_SIZE + ALAC_FRAME_FOOTER_SIZE +
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388 avctx->frame_size*avctx->channels*avctx->bits_per_coded_sample)>>3;
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389
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390 s->write_sample_size = avctx->bits_per_coded_sample + avctx->channels - 1; // FIXME: consider wasted_bytes
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391
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392 AV_WB32(alac_extradata, ALAC_EXTRADATA_SIZE);
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393 AV_WB32(alac_extradata+4, MKBETAG('a','l','a','c'));
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394 AV_WB32(alac_extradata+12, avctx->frame_size);
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395 AV_WB8 (alac_extradata+17, avctx->bits_per_coded_sample);
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396 AV_WB8 (alac_extradata+21, avctx->channels);
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397 AV_WB32(alac_extradata+24, s->max_coded_frame_size);
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398 AV_WB32(alac_extradata+28, avctx->sample_rate*avctx->channels*avctx->bits_per_coded_sample); // average bitrate
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399 AV_WB32(alac_extradata+32, avctx->sample_rate);
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400
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401 // Set relevant extradata fields
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402 if(s->compression_level > 0) {
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403 AV_WB8(alac_extradata+18, s->rc.history_mult);
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404 AV_WB8(alac_extradata+19, s->rc.initial_history);
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405 AV_WB8(alac_extradata+20, s->rc.k_modifier);
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406 }
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407
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0753d03d232a alacenc: last few hunks approved by michael
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408 s->min_prediction_order = DEFAULT_MIN_PRED_ORDER;
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409 if(avctx->min_prediction_order >= 0) {
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410 if(avctx->min_prediction_order < MIN_LPC_ORDER ||
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411 avctx->min_prediction_order > ALAC_MAX_LPC_ORDER) {
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412 av_log(avctx, AV_LOG_ERROR, "invalid min prediction order: %d\n", avctx->min_prediction_order);
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413 return -1;
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414 }
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415
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416 s->min_prediction_order = avctx->min_prediction_order;
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417 }
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418
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419 s->max_prediction_order = DEFAULT_MAX_PRED_ORDER;
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420 if(avctx->max_prediction_order >= 0) {
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421 if(avctx->max_prediction_order < MIN_LPC_ORDER ||
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422 avctx->max_prediction_order > ALAC_MAX_LPC_ORDER) {
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423 av_log(avctx, AV_LOG_ERROR, "invalid max prediction order: %d\n", avctx->max_prediction_order);
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424 return -1;
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425 }
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426
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427 s->max_prediction_order = avctx->max_prediction_order;
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428 }
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429
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430 if(s->max_prediction_order < s->min_prediction_order) {
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431 av_log(avctx, AV_LOG_ERROR, "invalid prediction orders: min=%d max=%d\n",
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432 s->min_prediction_order, s->max_prediction_order);
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433 return -1;
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434 }
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435
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436 avctx->extradata = alac_extradata;
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437 avctx->extradata_size = ALAC_EXTRADATA_SIZE;
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438
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439 avctx->coded_frame = avcodec_alloc_frame();
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440 avctx->coded_frame->key_frame = 1;
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441
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442 s->avctx = avctx;
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443 dsputil_init(&s->dspctx, avctx);
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444
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445 return 0;
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446 }
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447
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448 static int alac_encode_frame(AVCodecContext *avctx, uint8_t *frame,
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449 int buf_size, void *data)
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450 {
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451 AlacEncodeContext *s = avctx->priv_data;
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452 PutBitContext *pb = &s->pbctx;
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453 int i, out_bytes, verbatim_flag = 0;
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454
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455 if(avctx->frame_size > DEFAULT_FRAME_SIZE) {
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456 av_log(avctx, AV_LOG_ERROR, "input frame size exceeded\n");
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457 return -1;
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458 }
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459
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460 if(buf_size < 2*s->max_coded_frame_size) {
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461 av_log(avctx, AV_LOG_ERROR, "buffer size is too small\n");
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462 return -1;
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463 }
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464
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465 verbatim:
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466 init_put_bits(pb, frame, buf_size);
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467
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468 if((s->compression_level == 0) || verbatim_flag) {
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diff changeset
469 // Verbatim mode
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470 int16_t *samples = data;
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471 write_frame_header(s, 1);
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472 for(i=0; i<avctx->frame_size*avctx->channels; i++) {
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473 put_sbits(pb, 16, *samples++);
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474 }
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475 } else {
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476 init_sample_buffers(s, data);
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477 write_frame_header(s, 0);
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478 write_compressed_frame(s);
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479 }
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480
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481 put_bits(pb, 3, 7);
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482 flush_put_bits(pb);
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483 out_bytes = put_bits_count(pb) >> 3;
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484
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diff changeset
485 if(out_bytes > s->max_coded_frame_size) {
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parents: 7598
diff changeset
486 /* frame too large. use verbatim mode */
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diff changeset
487 if(verbatim_flag || (s->compression_level == 0)) {
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diff changeset
488 /* still too large. must be an error. */
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489 av_log(avctx, AV_LOG_ERROR, "error encoding frame\n");
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490 return -1;
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491 }
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492 verbatim_flag = 1;
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493 goto verbatim;
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494 }
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495
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496 return out_bytes;
7595
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parents:
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497 }
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parents:
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498
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499 static av_cold int alac_encode_close(AVCodecContext *avctx)
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500 {
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501 av_freep(&avctx->extradata);
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502 avctx->extradata_size = 0;
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503 av_freep(&avctx->coded_frame);
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504 return 0;
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parents:
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505 }
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parents:
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506
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parents:
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507 AVCodec alac_encoder = {
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parents:
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508 "alac",
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parents:
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509 CODEC_TYPE_AUDIO,
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parents:
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510 CODEC_ID_ALAC,
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parents:
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511 sizeof(AlacEncodeContext),
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parents:
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512 alac_encode_init,
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parents:
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513 alac_encode_frame,
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parents:
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514 alac_encode_close,
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parents:
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515 .capabilities = CODEC_CAP_SMALL_LAST_FRAME,
7598
60dd0089cdba alacenc: NULL_IF_CONFIG_SMALL long_name.
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516 .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
7595
30af3a9775de Import ok'd parts of ALAC encoder from GSoC repo.
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parents:
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517 };