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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
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37 int interlacing_shift;
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38 int interlacing_leftweight;
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39 PutBitContext pbctx;
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40 DSPContext dspctx;
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41 AVCodecContext *avctx;
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42 } AlacEncodeContext;
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43
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44
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45 static void encode_scalar(AlacEncodeContext *s, int x, int k, int write_sample_size)
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46 {
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47 int divisor, q, r;
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48
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49 k = FFMIN(k, s->rc.k_modifier);
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50 divisor = (1<<k) - 1;
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51 q = x / divisor;
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52 r = x % divisor;
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53
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54 if(q > 8) {
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55 // write escape code and sample value directly
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56 put_bits(&s->pbctx, 9, ALAC_ESCAPE_CODE);
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57 put_bits(&s->pbctx, write_sample_size, x);
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58 } else {
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59 if(q)
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60 put_bits(&s->pbctx, q, (1<<q) - 1);
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61 put_bits(&s->pbctx, 1, 0);
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62
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63 if(k != 1) {
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64 if(r > 0)
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65 put_bits(&s->pbctx, k, r+1);
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66 else
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67 put_bits(&s->pbctx, k-1, 0);
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68 }
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69 }
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70 }
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71
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72 static void write_frame_header(AlacEncodeContext *s, int is_verbatim)
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73 {
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74 put_bits(&s->pbctx, 3, s->channels-1); // No. of channels -1
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75 put_bits(&s->pbctx, 16, 0); // Seems to be zero
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76 put_bits(&s->pbctx, 1, 1); // Sample count is in the header
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77 put_bits(&s->pbctx, 2, 0); // FIXME: Wasted bytes field
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78 put_bits(&s->pbctx, 1, is_verbatim); // Audio block is verbatim
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79 put_bits(&s->pbctx, 32, s->avctx->frame_size); // No. of samples in the frame
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80 }
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81
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82 static void write_compressed_frame(AlacEncodeContext *s)
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83 {
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84 int i, j;
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85
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86 /* only simple mid/side decorrelation supported as of now */
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87 alac_stereo_decorrelation(s);
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88 put_bits(&s->pbctx, 8, s->interlacing_shift);
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89 put_bits(&s->pbctx, 8, s->interlacing_leftweight);
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90
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91 for(i=0;i<s->channels;i++) {
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92
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93 calc_predictor_params(s, i);
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94
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95 put_bits(&s->pbctx, 4, 0); // prediction type : currently only type 0 has been RE'd
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96 put_bits(&s->pbctx, 4, s->lpc[i].lpc_quant);
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97
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98 put_bits(&s->pbctx, 3, s->rc.rice_modifier);
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99 put_bits(&s->pbctx, 5, s->lpc[i].lpc_order);
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100 // predictor coeff. table
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101 for(j=0;j<s->lpc[i].lpc_order;j++) {
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102 put_sbits(&s->pbctx, 16, s->lpc[i].lpc_coeff[j]);
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103 }
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104 }
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105
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106 // apply lpc and entropy coding to audio samples
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107
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108 for(i=0;i<s->channels;i++) {
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109 alac_linear_predictor(s, i);
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110 alac_entropy_coder(s);
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111 }
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112 }
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113
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114 static av_cold int alac_encode_init(AVCodecContext *avctx)
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115 {
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116 AlacEncodeContext *s = avctx->priv_data;
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117 uint8_t *alac_extradata = av_mallocz(ALAC_EXTRADATA_SIZE+1);
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118
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119 avctx->frame_size = DEFAULT_FRAME_SIZE;
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120 avctx->bits_per_sample = DEFAULT_SAMPLE_SIZE;
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121 s->channels = avctx->channels;
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122 s->samplerate = avctx->sample_rate;
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123
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124 if(avctx->sample_fmt != SAMPLE_FMT_S16) {
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125 av_log(avctx, AV_LOG_ERROR, "only pcm_s16 input samples are supported\n");
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126 return -1;
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127 }
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128
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129 // Set default compression level
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130 if(avctx->compression_level == FF_COMPRESSION_DEFAULT)
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131 s->compression_level = 1;
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132 else
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133 s->compression_level = av_clip(avctx->compression_level, 0, 1);
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134
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135 // Initialize default Rice parameters
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136 s->rc.history_mult = 40;
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137 s->rc.initial_history = 10;
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138 s->rc.k_modifier = 14;
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139 s->rc.rice_modifier = 4;
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140
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141 s->max_coded_frame_size = (ALAC_FRAME_HEADER_SIZE + ALAC_FRAME_FOOTER_SIZE +
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142 avctx->frame_size*s->channels*avctx->bits_per_sample)>>3;
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143
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144 s->write_sample_size = avctx->bits_per_sample + s->channels - 1; // FIXME: consider wasted_bytes
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145
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146 AV_WB32(alac_extradata, ALAC_EXTRADATA_SIZE);
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147 AV_WB32(alac_extradata+4, MKBETAG('a','l','a','c'));
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148 AV_WB32(alac_extradata+12, avctx->frame_size);
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149 AV_WB8 (alac_extradata+17, avctx->bits_per_sample);
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150 AV_WB8 (alac_extradata+21, s->channels);
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151 AV_WB32(alac_extradata+24, s->max_coded_frame_size);
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152 AV_WB32(alac_extradata+28, s->samplerate*s->channels*avctx->bits_per_sample); // average bitrate
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153 AV_WB32(alac_extradata+32, s->samplerate);
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154
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155 // Set relevant extradata fields
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156 if(s->compression_level > 0) {
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157 AV_WB8(alac_extradata+18, s->rc.history_mult);
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158 AV_WB8(alac_extradata+19, s->rc.initial_history);
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159 AV_WB8(alac_extradata+20, s->rc.k_modifier);
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160 }
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161
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162 avctx->extradata = alac_extradata;
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163 avctx->extradata_size = ALAC_EXTRADATA_SIZE;
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164
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165 avctx->coded_frame = avcodec_alloc_frame();
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166 avctx->coded_frame->key_frame = 1;
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167
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168 s->avctx = avctx;
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169 dsputil_init(&s->dspctx, avctx);
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170
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171 allocate_sample_buffers(s);
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172
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173 return 0;
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174 }
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175
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176 static av_cold int alac_encode_close(AVCodecContext *avctx)
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177 {
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178 AlacEncodeContext *s = avctx->priv_data;
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179
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180 av_freep(&avctx->extradata);
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181 avctx->extradata_size = 0;
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182 av_freep(&avctx->coded_frame);
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183 free_sample_buffers(s);
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184 return 0;
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185 }
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186
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187 AVCodec alac_encoder = {
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188 "alac",
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189 CODEC_TYPE_AUDIO,
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190 CODEC_ID_ALAC,
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191 sizeof(AlacEncodeContext),
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192 alac_encode_init,
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193 alac_encode_frame,
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194 alac_encode_close,
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195 .capabilities = CODEC_CAP_SMALL_LAST_FRAME,
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196 .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
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197 };
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