annotate roqaudioenc.c @ 11254:e820a9d3f80c libavcodec

Decode alpha plane in Bink video
author kostya
date Tue, 23 Feb 2010 07:07:54 +0000
parents 1a3908177e87
children 421c9441c11e
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1 /*
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2 * RoQ audio encoder
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3 *
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4 * Copyright (c) 2005 Eric Lasota
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5 * Based on RoQ specs (c)2001 Tim Ferguson
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6 *
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7 * This file is part of FFmpeg.
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8 *
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9 * FFmpeg is free software; you can redistribute it and/or
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10 * modify it under the terms of the GNU Lesser General Public
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11 * License as published by the Free Software Foundation; either
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12 * version 2.1 of the License, or (at your option) any later version.
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13 *
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14 * FFmpeg is distributed in the hope that it will be useful,
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15 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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16 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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17 * Lesser General Public License for more details.
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18 *
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19 * You should have received a copy of the GNU Lesser General Public
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20 * License along with FFmpeg; if not, write to the Free Software
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21 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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22 */
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23
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24 #include "avcodec.h"
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25 #include "bytestream.h"
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26
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27 #define ROQ_FIRST_FRAME_SIZE (735*8)
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28 #define ROQ_FRAME_SIZE 735
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29
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30
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31 #define MAX_DPCM (127*127)
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32
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33
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34 typedef struct
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35 {
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36 short lastSample[2];
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37 } ROQDPCMContext;
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38
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39 static av_cold int roq_dpcm_encode_init(AVCodecContext *avctx)
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40 {
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41 ROQDPCMContext *context = avctx->priv_data;
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42
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43 if (avctx->channels > 2) {
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44 av_log(avctx, AV_LOG_ERROR, "Audio must be mono or stereo\n");
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45 return -1;
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46 }
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47 if (avctx->sample_rate != 22050) {
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48 av_log(avctx, AV_LOG_ERROR, "Audio must be 22050 Hz\n");
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49 return -1;
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50 }
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51 if (avctx->sample_fmt != SAMPLE_FMT_S16) {
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52 av_log(avctx, AV_LOG_ERROR, "Audio must be signed 16-bit\n");
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53 return -1;
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54 }
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55
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56 avctx->frame_size = ROQ_FIRST_FRAME_SIZE;
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57
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58 context->lastSample[0] = context->lastSample[1] = 0;
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59
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60 avctx->coded_frame= avcodec_alloc_frame();
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61 avctx->coded_frame->key_frame= 1;
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62
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63 return 0;
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64 }
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65
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66 static unsigned char dpcm_predict(short *previous, short current)
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67 {
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68 int diff;
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69 int negative;
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70 int result;
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71 int predicted;
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72
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73 diff = current - *previous;
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74
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75 negative = diff<0;
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76 diff = FFABS(diff);
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77
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78 if (diff >= MAX_DPCM)
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79 result = 127;
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80 else {
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81 result = ff_sqrt(diff);
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82 result += diff > result*result+result;
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83 }
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84
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85 /* See if this overflows */
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86 retry:
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87 diff = result*result;
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88 if (negative)
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89 diff = -diff;
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90 predicted = *previous + diff;
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91
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92 /* If it overflows, back off a step */
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93 if (predicted > 32767 || predicted < -32768) {
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94 result--;
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95 goto retry;
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96 }
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97
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98 /* Add the sign bit */
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99 result |= negative << 7; //if (negative) result |= 128;
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100
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101 *previous = predicted;
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102
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103 return result;
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104 }
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105
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106 static int roq_dpcm_encode_frame(AVCodecContext *avctx,
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107 unsigned char *frame, int buf_size, void *data)
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108 {
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109 int i, samples, stereo, ch;
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110 short *in;
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111 unsigned char *out;
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112
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113 ROQDPCMContext *context = avctx->priv_data;
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114
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115 stereo = (avctx->channels == 2);
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116
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117 if (stereo) {
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118 context->lastSample[0] &= 0xFF00;
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119 context->lastSample[1] &= 0xFF00;
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120 }
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121
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122 out = frame;
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123 in = data;
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124
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125 bytestream_put_byte(&out, stereo ? 0x21 : 0x20);
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126 bytestream_put_byte(&out, 0x10);
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127 bytestream_put_le32(&out, avctx->frame_size*avctx->channels);
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128
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129 if (stereo) {
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130 bytestream_put_byte(&out, (context->lastSample[1])>>8);
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131 bytestream_put_byte(&out, (context->lastSample[0])>>8);
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132 } else
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133 bytestream_put_le16(&out, context->lastSample[0]);
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134
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135 /* Write the actual samples */
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136 samples = avctx->frame_size;
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137 for (i=0; i<samples; i++)
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138 for (ch=0; ch<avctx->channels; ch++)
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139 *out++ = dpcm_predict(&context->lastSample[ch], *in++);
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140
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141 /* Use smaller frames from now on */
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142 avctx->frame_size = ROQ_FRAME_SIZE;
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143
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144 /* Return the result size */
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145 return out - frame;
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146 }
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147
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148 static av_cold int roq_dpcm_encode_close(AVCodecContext *avctx)
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149 {
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150 av_freep(&avctx->coded_frame);
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151
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152 return 0;
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153 }
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154
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155 AVCodec roq_dpcm_encoder = {
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156 "roq_dpcm",
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157 CODEC_TYPE_AUDIO,
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158 CODEC_ID_ROQ_DPCM,
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159 sizeof(ROQDPCMContext),
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160 roq_dpcm_encode_init,
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161 roq_dpcm_encode_frame,
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162 roq_dpcm_encode_close,
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163 NULL,
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164 .sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
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165 .long_name = NULL_IF_CONFIG_SMALL("id RoQ DPCM"),
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166 };