annotate alacenc.c @ 9807:08b84fcd478c libavcodec

smacker video decoder uses reget_buffer, set CODEC_CAP_DR1
author bcoudurier
date Fri, 05 Jun 2009 08:37:55 +0000
parents 0dce4fe6e6f3
children f55ca9a2b948
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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 "get_bits.h"
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24 #include "put_bits.h"
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25 #include "dsputil.h"
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26 #include "lpc.h"
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27 #include "mathops.h"
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28
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29 #define DEFAULT_FRAME_SIZE 4096
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30 #define DEFAULT_SAMPLE_SIZE 16
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31 #define MAX_CHANNELS 8
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32 #define ALAC_EXTRADATA_SIZE 36
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33 #define ALAC_FRAME_HEADER_SIZE 55
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34 #define ALAC_FRAME_FOOTER_SIZE 3
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35
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36 #define ALAC_ESCAPE_CODE 0x1FF
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37 #define ALAC_MAX_LPC_ORDER 30
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38 #define DEFAULT_MAX_PRED_ORDER 6
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39 #define DEFAULT_MIN_PRED_ORDER 4
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40 #define ALAC_MAX_LPC_PRECISION 9
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41 #define ALAC_MAX_LPC_SHIFT 9
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42
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43 #define ALAC_CHMODE_LEFT_RIGHT 0
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44 #define ALAC_CHMODE_LEFT_SIDE 1
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45 #define ALAC_CHMODE_RIGHT_SIDE 2
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46 #define ALAC_CHMODE_MID_SIDE 3
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47
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48 typedef struct RiceContext {
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49 int history_mult;
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50 int initial_history;
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51 int k_modifier;
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52 int rice_modifier;
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53 } RiceContext;
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54
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55 typedef struct LPCContext {
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56 int lpc_order;
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57 int lpc_coeff[ALAC_MAX_LPC_ORDER+1];
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58 int lpc_quant;
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59 } LPCContext;
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60
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61 typedef struct AlacEncodeContext {
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62 int compression_level;
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63 int min_prediction_order;
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64 int max_prediction_order;
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65 int max_coded_frame_size;
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66 int write_sample_size;
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67 int32_t sample_buf[MAX_CHANNELS][DEFAULT_FRAME_SIZE];
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68 int32_t predictor_buf[DEFAULT_FRAME_SIZE];
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69 int interlacing_shift;
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70 int interlacing_leftweight;
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71 PutBitContext pbctx;
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72 RiceContext rc;
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73 LPCContext lpc[MAX_CHANNELS];
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74 DSPContext dspctx;
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75 AVCodecContext *avctx;
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76 } AlacEncodeContext;
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77
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78
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79 static void init_sample_buffers(AlacEncodeContext *s, int16_t *input_samples)
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80 {
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81 int ch, i;
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82
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83 for(ch=0;ch<s->avctx->channels;ch++) {
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84 int16_t *sptr = input_samples + ch;
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85 for(i=0;i<s->avctx->frame_size;i++) {
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86 s->sample_buf[ch][i] = *sptr;
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87 sptr += s->avctx->channels;
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88 }
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89 }
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90 }
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91
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92 static void encode_scalar(AlacEncodeContext *s, int x, int k, int write_sample_size)
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93 {
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94 int divisor, q, r;
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95
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96 k = FFMIN(k, s->rc.k_modifier);
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97 divisor = (1<<k) - 1;
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98 q = x / divisor;
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99 r = x % divisor;
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100
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101 if(q > 8) {
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102 // write escape code and sample value directly
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103 put_bits(&s->pbctx, 9, ALAC_ESCAPE_CODE);
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104 put_bits(&s->pbctx, write_sample_size, x);
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105 } else {
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106 if(q)
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107 put_bits(&s->pbctx, q, (1<<q) - 1);
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108 put_bits(&s->pbctx, 1, 0);
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109
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110 if(k != 1) {
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111 if(r > 0)
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112 put_bits(&s->pbctx, k, r+1);
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113 else
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114 put_bits(&s->pbctx, k-1, 0);
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115 }
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116 }
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117 }
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118
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119 static void write_frame_header(AlacEncodeContext *s, int is_verbatim)
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120 {
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121 put_bits(&s->pbctx, 3, s->avctx->channels-1); // No. of channels -1
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122 put_bits(&s->pbctx, 16, 0); // Seems to be zero
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123 put_bits(&s->pbctx, 1, 1); // Sample count is in the header
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124 put_bits(&s->pbctx, 2, 0); // FIXME: Wasted bytes field
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125 put_bits(&s->pbctx, 1, is_verbatim); // Audio block is verbatim
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126 put_bits(&s->pbctx, 32, s->avctx->frame_size); // No. of samples in the frame
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127 }
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128
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129 static void calc_predictor_params(AlacEncodeContext *s, int ch)
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130 {
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131 int32_t coefs[MAX_LPC_ORDER][MAX_LPC_ORDER];
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132 int shift[MAX_LPC_ORDER];
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133 int opt_order;
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134
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135 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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136 ALAC_MAX_LPC_PRECISION, coefs, shift, 1, ORDER_METHOD_EST, ALAC_MAX_LPC_SHIFT, 1);
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137
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138 s->lpc[ch].lpc_order = opt_order;
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139 s->lpc[ch].lpc_quant = shift[opt_order-1];
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140 memcpy(s->lpc[ch].lpc_coeff, coefs[opt_order-1], opt_order*sizeof(int));
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141 }
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142
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143 static int estimate_stereo_mode(int32_t *left_ch, int32_t *right_ch, int n)
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144 {
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145 int i, best;
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146 int32_t lt, rt;
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147 uint64_t sum[4];
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148 uint64_t score[4];
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149
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150 /* calculate sum of 2nd order residual for each channel */
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151 sum[0] = sum[1] = sum[2] = sum[3] = 0;
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152 for(i=2; i<n; i++) {
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153 lt = left_ch[i] - 2*left_ch[i-1] + left_ch[i-2];
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154 rt = right_ch[i] - 2*right_ch[i-1] + right_ch[i-2];
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155 sum[2] += FFABS((lt + rt) >> 1);
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156 sum[3] += FFABS(lt - rt);
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157 sum[0] += FFABS(lt);
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158 sum[1] += FFABS(rt);
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159 }
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160
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161 /* calculate score for each mode */
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162 score[0] = sum[0] + sum[1];
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163 score[1] = sum[0] + sum[3];
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164 score[2] = sum[1] + sum[3];
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165 score[3] = sum[2] + sum[3];
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166
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167 /* return mode with lowest score */
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168 best = 0;
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169 for(i=1; i<4; i++) {
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170 if(score[i] < score[best]) {
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171 best = i;
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172 }
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173 }
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174 return best;
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175 }
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176
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177 static void alac_stereo_decorrelation(AlacEncodeContext *s)
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178 {
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179 int32_t *left = s->sample_buf[0], *right = s->sample_buf[1];
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180 int i, mode, n = s->avctx->frame_size;
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181 int32_t tmp;
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182
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183 mode = estimate_stereo_mode(left, right, n);
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184
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185 switch(mode)
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186 {
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187 case ALAC_CHMODE_LEFT_RIGHT:
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188 s->interlacing_leftweight = 0;
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189 s->interlacing_shift = 0;
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190 break;
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191
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192 case ALAC_CHMODE_LEFT_SIDE:
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193 for(i=0; i<n; i++) {
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194 right[i] = left[i] - right[i];
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195 }
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196 s->interlacing_leftweight = 1;
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197 s->interlacing_shift = 0;
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198 break;
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199
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200 case ALAC_CHMODE_RIGHT_SIDE:
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201 for(i=0; i<n; i++) {
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202 tmp = right[i];
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203 right[i] = left[i] - right[i];
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204 left[i] = tmp + (right[i] >> 31);
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205 }
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206 s->interlacing_leftweight = 1;
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207 s->interlacing_shift = 31;
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208 break;
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209
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210 default:
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211 for(i=0; i<n; i++) {
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212 tmp = left[i];
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213 left[i] = (tmp + right[i]) >> 1;
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214 right[i] = tmp - right[i];
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215 }
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216 s->interlacing_leftweight = 1;
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217 s->interlacing_shift = 1;
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218 break;
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219 }
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220 }
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221
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222 static void alac_linear_predictor(AlacEncodeContext *s, int ch)
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223 {
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224 int i;
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225 LPCContext lpc = s->lpc[ch];
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226
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227 if(lpc.lpc_order == 31) {
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228 s->predictor_buf[0] = s->sample_buf[ch][0];
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229
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230 for(i=1; i<s->avctx->frame_size; i++)
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231 s->predictor_buf[i] = s->sample_buf[ch][i] - s->sample_buf[ch][i-1];
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232
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233 return;
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234 }
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235
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236 // generalised linear predictor
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237
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238 if(lpc.lpc_order > 0) {
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239 int32_t *samples = s->sample_buf[ch];
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240 int32_t *residual = s->predictor_buf;
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241
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242 // generate warm-up samples
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243 residual[0] = samples[0];
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244 for(i=1;i<=lpc.lpc_order;i++)
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245 residual[i] = samples[i] - samples[i-1];
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246
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247 // perform lpc on remaining samples
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248 for(i = lpc.lpc_order + 1; i < s->avctx->frame_size; i++) {
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249 int sum = 1 << (lpc.lpc_quant - 1), res_val, j;
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250
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251 for (j = 0; j < lpc.lpc_order; j++) {
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252 sum += (samples[lpc.lpc_order-j] - samples[0]) *
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253 lpc.lpc_coeff[j];
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254 }
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255
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256 sum >>= lpc.lpc_quant;
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257 sum += samples[0];
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258 residual[i] = sign_extend(samples[lpc.lpc_order+1] - sum,
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259 s->write_sample_size);
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260 res_val = residual[i];
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261
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262 if(res_val) {
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263 int index = lpc.lpc_order - 1;
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264 int neg = (res_val < 0);
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265
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266 while(index >= 0 && (neg ? (res_val < 0):(res_val > 0))) {
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267 int val = samples[0] - samples[lpc.lpc_order - index];
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268 int sign = (val ? FFSIGN(val) : 0);
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269
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270 if(neg)
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271 sign*=-1;
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272
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273 lpc.lpc_coeff[index] -= sign;
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274 val *= sign;
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275 res_val -= ((val >> lpc.lpc_quant) *
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276 (lpc.lpc_order - index));
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277 index--;
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278 }
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279 }
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280 samples++;
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281 }
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282 }
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283 }
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284
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285 static void alac_entropy_coder(AlacEncodeContext *s)
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286 {
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287 unsigned int history = s->rc.initial_history;
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288 int sign_modifier = 0, i, k;
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289 int32_t *samples = s->predictor_buf;
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290
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291 for(i=0;i < s->avctx->frame_size;) {
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292 int x;
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293
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294 k = av_log2((history >> 9) + 3);
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295
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296 x = -2*(*samples)-1;
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297 x ^= (x>>31);
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298
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299 samples++;
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300 i++;
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301
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302 encode_scalar(s, x - sign_modifier, k, s->write_sample_size);
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303
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304 history += x * s->rc.history_mult
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305 - ((history * s->rc.history_mult) >> 9);
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306
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307 sign_modifier = 0;
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308 if(x > 0xFFFF)
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309 history = 0xFFFF;
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310
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311 if((history < 128) && (i < s->avctx->frame_size)) {
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312 unsigned int block_size = 0;
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313
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314 k = 7 - av_log2(history) + ((history + 16) >> 6);
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315
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316 while((*samples == 0) && (i < s->avctx->frame_size)) {
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317 samples++;
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318 i++;
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319 block_size++;
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320 }
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321 encode_scalar(s, block_size, k, 16);
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322
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323 sign_modifier = (block_size <= 0xFFFF);
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324
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325 history = 0;
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326 }
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327
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diff changeset
328 }
0753d03d232a alacenc: last few hunks approved by michael
jai_menon
parents: 7618
diff changeset
329 }
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jai_menon
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diff changeset
330
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331 static void write_compressed_frame(AlacEncodeContext *s)
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332 {
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333 int i, j;
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334
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b87a9296e854 alacenc : perform decorrelation only for stereo samples
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335 if(s->avctx->channels == 2)
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336 alac_stereo_decorrelation(s);
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337 put_bits(&s->pbctx, 8, s->interlacing_shift);
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338 put_bits(&s->pbctx, 8, s->interlacing_leftweight);
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339
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340 for(i=0;i<s->avctx->channels;i++) {
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341
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342 calc_predictor_params(s, i);
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343
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344 put_bits(&s->pbctx, 4, 0); // prediction type : currently only type 0 has been RE'd
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345 put_bits(&s->pbctx, 4, s->lpc[i].lpc_quant);
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346
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347 put_bits(&s->pbctx, 3, s->rc.rice_modifier);
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348 put_bits(&s->pbctx, 5, s->lpc[i].lpc_order);
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349 // predictor coeff. table
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350 for(j=0;j<s->lpc[i].lpc_order;j++) {
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351 put_sbits(&s->pbctx, 16, s->lpc[i].lpc_coeff[j]);
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352 }
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353 }
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354
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355 // apply lpc and entropy coding to audio samples
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356
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357 for(i=0;i<s->avctx->channels;i++) {
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358 alac_linear_predictor(s, i);
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359 alac_entropy_coder(s);
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360 }
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361 }
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362
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363 static av_cold int alac_encode_init(AVCodecContext *avctx)
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364 {
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365 AlacEncodeContext *s = avctx->priv_data;
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366 uint8_t *alac_extradata = av_mallocz(ALAC_EXTRADATA_SIZE+1);
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367
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368 avctx->frame_size = DEFAULT_FRAME_SIZE;
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369 avctx->bits_per_coded_sample = DEFAULT_SAMPLE_SIZE;
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370
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371 if(avctx->sample_fmt != SAMPLE_FMT_S16) {
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372 av_log(avctx, AV_LOG_ERROR, "only pcm_s16 input samples are supported\n");
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373 return -1;
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374 }
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375
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ramiro
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376 // Set default compression level
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377 if(avctx->compression_level == FF_COMPRESSION_DEFAULT)
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378 s->compression_level = 1;
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379 else
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380 s->compression_level = av_clip(avctx->compression_level, 0, 1);
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381
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382 // Initialize default Rice parameters
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383 s->rc.history_mult = 40;
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384 s->rc.initial_history = 10;
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385 s->rc.k_modifier = 14;
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386 s->rc.rice_modifier = 4;
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387
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388 s->max_coded_frame_size = (ALAC_FRAME_HEADER_SIZE + ALAC_FRAME_FOOTER_SIZE +
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4525dcd81357 Bump Major version, this commit is almost just renaming bits_per_sample to
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diff changeset
389 avctx->frame_size*avctx->channels*avctx->bits_per_coded_sample)>>3;
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390
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391 s->write_sample_size = avctx->bits_per_coded_sample + avctx->channels - 1; // FIXME: consider wasted_bytes
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392
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393 AV_WB32(alac_extradata, ALAC_EXTRADATA_SIZE);
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394 AV_WB32(alac_extradata+4, MKBETAG('a','l','a','c'));
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395 AV_WB32(alac_extradata+12, avctx->frame_size);
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396 AV_WB8 (alac_extradata+17, avctx->bits_per_coded_sample);
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397 AV_WB8 (alac_extradata+21, avctx->channels);
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398 AV_WB32(alac_extradata+24, s->max_coded_frame_size);
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diff changeset
399 AV_WB32(alac_extradata+28, avctx->sample_rate*avctx->channels*avctx->bits_per_coded_sample); // average bitrate
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400 AV_WB32(alac_extradata+32, avctx->sample_rate);
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401
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diff changeset
402 // Set relevant extradata fields
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403 if(s->compression_level > 0) {
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parents:
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404 AV_WB8(alac_extradata+18, s->rc.history_mult);
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parents:
diff changeset
405 AV_WB8(alac_extradata+19, s->rc.initial_history);
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parents:
diff changeset
406 AV_WB8(alac_extradata+20, s->rc.k_modifier);
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parents:
diff changeset
407 }
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ramiro
parents:
diff changeset
408
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0753d03d232a alacenc: last few hunks approved by michael
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409 s->min_prediction_order = DEFAULT_MIN_PRED_ORDER;
7617
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diff changeset
410 if(avctx->min_prediction_order >= 0) {
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diff changeset
411 if(avctx->min_prediction_order < MIN_LPC_ORDER ||
7620
5c2299115d1f alacenc: compare against ALAC_MAX_LPC_ORDER instead of MAX_LPC_ORDER
jai_menon
parents: 7619
diff changeset
412 avctx->min_prediction_order > ALAC_MAX_LPC_ORDER) {
7617
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413 av_log(avctx, AV_LOG_ERROR, "invalid min prediction order: %d\n", avctx->min_prediction_order);
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414 return -1;
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diff changeset
415 }
924dc060db81 Import more OKed parts of ALAC encoder from GSoC repo.
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diff changeset
416
924dc060db81 Import more OKed parts of ALAC encoder from GSoC repo.
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diff changeset
417 s->min_prediction_order = avctx->min_prediction_order;
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diff changeset
418 }
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diff changeset
419
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jai_menon
parents: 7618
diff changeset
420 s->max_prediction_order = DEFAULT_MAX_PRED_ORDER;
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diff changeset
421 if(avctx->max_prediction_order >= 0) {
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diff changeset
422 if(avctx->max_prediction_order < MIN_LPC_ORDER ||
7620
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jai_menon
parents: 7619
diff changeset
423 avctx->max_prediction_order > ALAC_MAX_LPC_ORDER) {
7617
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424 av_log(avctx, AV_LOG_ERROR, "invalid max prediction order: %d\n", avctx->max_prediction_order);
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425 return -1;
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diff changeset
426 }
924dc060db81 Import more OKed parts of ALAC encoder from GSoC repo.
ramiro
parents: 7604
diff changeset
427
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diff changeset
428 s->max_prediction_order = avctx->max_prediction_order;
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diff changeset
429 }
924dc060db81 Import more OKed parts of ALAC encoder from GSoC repo.
ramiro
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diff changeset
430
924dc060db81 Import more OKed parts of ALAC encoder from GSoC repo.
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parents: 7604
diff changeset
431 if(s->max_prediction_order < s->min_prediction_order) {
924dc060db81 Import more OKed parts of ALAC encoder from GSoC repo.
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parents: 7604
diff changeset
432 av_log(avctx, AV_LOG_ERROR, "invalid prediction orders: min=%d max=%d\n",
924dc060db81 Import more OKed parts of ALAC encoder from GSoC repo.
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diff changeset
433 s->min_prediction_order, s->max_prediction_order);
924dc060db81 Import more OKed parts of ALAC encoder from GSoC repo.
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diff changeset
434 return -1;
924dc060db81 Import more OKed parts of ALAC encoder from GSoC repo.
ramiro
parents: 7604
diff changeset
435 }
924dc060db81 Import more OKed parts of ALAC encoder from GSoC repo.
ramiro
parents: 7604
diff changeset
436
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diff changeset
437 avctx->extradata = alac_extradata;
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ramiro
parents:
diff changeset
438 avctx->extradata_size = ALAC_EXTRADATA_SIZE;
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ramiro
parents:
diff changeset
439
30af3a9775de Import ok'd parts of ALAC encoder from GSoC repo.
ramiro
parents:
diff changeset
440 avctx->coded_frame = avcodec_alloc_frame();
30af3a9775de Import ok'd parts of ALAC encoder from GSoC repo.
ramiro
parents:
diff changeset
441 avctx->coded_frame->key_frame = 1;
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ramiro
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diff changeset
442
30af3a9775de Import ok'd parts of ALAC encoder from GSoC repo.
ramiro
parents:
diff changeset
443 s->avctx = avctx;
30af3a9775de Import ok'd parts of ALAC encoder from GSoC repo.
ramiro
parents:
diff changeset
444 dsputil_init(&s->dspctx, avctx);
30af3a9775de Import ok'd parts of ALAC encoder from GSoC repo.
ramiro
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diff changeset
445
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6250ff63990b Import more ok'd parts of ALAC encoder from GSoC repo.
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diff changeset
446 return 0;
6250ff63990b Import more ok'd parts of ALAC encoder from GSoC repo.
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diff changeset
447 }
6250ff63990b Import more ok'd parts of ALAC encoder from GSoC repo.
ramiro
parents: 7598
diff changeset
448
6250ff63990b Import more ok'd parts of ALAC encoder from GSoC repo.
ramiro
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diff changeset
449 static int alac_encode_frame(AVCodecContext *avctx, uint8_t *frame,
6250ff63990b Import more ok'd parts of ALAC encoder from GSoC repo.
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diff changeset
450 int buf_size, void *data)
6250ff63990b Import more ok'd parts of ALAC encoder from GSoC repo.
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diff changeset
451 {
6250ff63990b Import more ok'd parts of ALAC encoder from GSoC repo.
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diff changeset
452 AlacEncodeContext *s = avctx->priv_data;
6250ff63990b Import more ok'd parts of ALAC encoder from GSoC repo.
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diff changeset
453 PutBitContext *pb = &s->pbctx;
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diff changeset
454 int i, out_bytes, verbatim_flag = 0;
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diff changeset
455
6250ff63990b Import more ok'd parts of ALAC encoder from GSoC repo.
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diff changeset
456 if(avctx->frame_size > DEFAULT_FRAME_SIZE) {
6250ff63990b Import more ok'd parts of ALAC encoder from GSoC repo.
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diff changeset
457 av_log(avctx, AV_LOG_ERROR, "input frame size exceeded\n");
6250ff63990b Import more ok'd parts of ALAC encoder from GSoC repo.
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diff changeset
458 return -1;
6250ff63990b Import more ok'd parts of ALAC encoder from GSoC repo.
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diff changeset
459 }
6250ff63990b Import more ok'd parts of ALAC encoder from GSoC repo.
ramiro
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diff changeset
460
6250ff63990b Import more ok'd parts of ALAC encoder from GSoC repo.
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parents: 7598
diff changeset
461 if(buf_size < 2*s->max_coded_frame_size) {
6250ff63990b Import more ok'd parts of ALAC encoder from GSoC repo.
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diff changeset
462 av_log(avctx, AV_LOG_ERROR, "buffer size is too small\n");
6250ff63990b Import more ok'd parts of ALAC encoder from GSoC repo.
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diff changeset
463 return -1;
6250ff63990b Import more ok'd parts of ALAC encoder from GSoC repo.
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diff changeset
464 }
7595
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diff changeset
465
7617
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diff changeset
466 verbatim:
924dc060db81 Import more OKed parts of ALAC encoder from GSoC repo.
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diff changeset
467 init_put_bits(pb, frame, buf_size);
924dc060db81 Import more OKed parts of ALAC encoder from GSoC repo.
ramiro
parents: 7604
diff changeset
468
7604
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diff changeset
469 if((s->compression_level == 0) || verbatim_flag) {
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ramiro
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diff changeset
470 // Verbatim mode
6250ff63990b Import more ok'd parts of ALAC encoder from GSoC repo.
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diff changeset
471 int16_t *samples = data;
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diff changeset
472 write_frame_header(s, 1);
6250ff63990b Import more ok'd parts of ALAC encoder from GSoC repo.
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diff changeset
473 for(i=0; i<avctx->frame_size*avctx->channels; i++) {
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diff changeset
474 put_sbits(pb, 16, *samples++);
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diff changeset
475 }
6250ff63990b Import more ok'd parts of ALAC encoder from GSoC repo.
ramiro
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diff changeset
476 } else {
6250ff63990b Import more ok'd parts of ALAC encoder from GSoC repo.
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diff changeset
477 init_sample_buffers(s, data);
6250ff63990b Import more ok'd parts of ALAC encoder from GSoC repo.
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diff changeset
478 write_frame_header(s, 0);
6250ff63990b Import more ok'd parts of ALAC encoder from GSoC repo.
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diff changeset
479 write_compressed_frame(s);
6250ff63990b Import more ok'd parts of ALAC encoder from GSoC repo.
ramiro
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diff changeset
480 }
6250ff63990b Import more ok'd parts of ALAC encoder from GSoC repo.
ramiro
parents: 7598
diff changeset
481
6250ff63990b Import more ok'd parts of ALAC encoder from GSoC repo.
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diff changeset
482 put_bits(pb, 3, 7);
6250ff63990b Import more ok'd parts of ALAC encoder from GSoC repo.
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diff changeset
483 flush_put_bits(pb);
6250ff63990b Import more ok'd parts of ALAC encoder from GSoC repo.
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diff changeset
484 out_bytes = put_bits_count(pb) >> 3;
6250ff63990b Import more ok'd parts of ALAC encoder from GSoC repo.
ramiro
parents: 7598
diff changeset
485
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ramiro
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486 if(out_bytes > s->max_coded_frame_size) {
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487 /* frame too large. use verbatim mode */
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488 if(verbatim_flag || (s->compression_level == 0)) {
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489 /* still too large. must be an error. */
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490 av_log(avctx, AV_LOG_ERROR, "error encoding frame\n");
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491 return -1;
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492 }
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493 verbatim_flag = 1;
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494 goto verbatim;
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495 }
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496
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497 return out_bytes;
7595
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498 }
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499
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500 static av_cold int alac_encode_close(AVCodecContext *avctx)
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501 {
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502 av_freep(&avctx->extradata);
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503 avctx->extradata_size = 0;
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504 av_freep(&avctx->coded_frame);
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505 return 0;
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506 }
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507
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508 AVCodec alac_encoder = {
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509 "alac",
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510 CODEC_TYPE_AUDIO,
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511 CODEC_ID_ALAC,
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512 sizeof(AlacEncodeContext),
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513 alac_encode_init,
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514 alac_encode_frame,
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515 alac_encode_close,
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516 .capabilities = CODEC_CAP_SMALL_LAST_FRAME,
7598
60dd0089cdba alacenc: NULL_IF_CONFIG_SMALL long_name.
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517 .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
7595
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518 };