annotate libmpcodecs/ad_msadpcm.c @ 34697:ac6b38cd0d45

Rename sub window video window. It was a bad idea to name the video window "sub window" at the time the GUI was written. The term "sub window" does make sense from the programmer's point of view, but it doesn't make any sense at all from the user's point of view, because the sub window simply is the window where the video will be displayed. Moreover, since the term "sub" is generally short for "subtitles", the renaming makes the code much easier to understand.
author ib
date Sat, 03 Mar 2012 16:45:15 +0000
parents a93891202051
children
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rev   line source
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1 /*
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2 * MS ADPCM decoder
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3 *
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4 * This file is responsible for decoding Microsoft ADPCM data.
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5 * Details about the data format can be found here:
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6 * http://www.pcisys.net/~melanson/codecs/
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7 *
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8 * Copyright (c) 2002 Mike Melanson
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9 *
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10 * This file is part of MPlayer.
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11 *
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12 * MPlayer is free software; you can redistribute it and/or modify
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13 * it under the terms of the GNU General Public License as published by
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14 * the Free Software Foundation; either version 2 of the License, or
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15 * (at your option) any later version.
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16 *
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17 * MPlayer is distributed in the hope that it will be useful,
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18 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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19 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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20 * GNU General Public License for more details.
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21 *
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22 * You should have received a copy of the GNU General Public License along
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23 * with MPlayer; if not, write to the Free Software Foundation, Inc.,
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24 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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25 */
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26
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27 #include <stdio.h>
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28 #include <stdlib.h>
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29 #include <unistd.h>
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30
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31 #include "config.h"
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32 #include "mp_msg.h"
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33 #include "libavutil/common.h"
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34 #include "libavutil/intreadwrite.h"
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35 #include "mpbswap.h"
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36 #include "ad_internal.h"
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37
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38 static const ad_info_t info =
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39 {
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40 "MS ADPCM audio decoder",
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41 "msadpcm",
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42 "Nick Kurshev",
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43 "Mike Melanson",
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44 ""
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45 };
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46
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47 LIBAD_EXTERN(msadpcm)
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48
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49 static const int ms_adapt_table[] =
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50 {
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51 230, 230, 230, 230, 307, 409, 512, 614,
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52 768, 614, 512, 409, 307, 230, 230, 230
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53 };
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54
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55 static const uint8_t ms_adapt_coeff1[] =
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56 {
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57 64, 128, 0, 48, 60, 115, 98
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58 };
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59
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60 static const int8_t ms_adapt_coeff2[] =
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61 {
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62 0, -64, 0, 16, 0, -52, -58
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63 };
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64
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65 #define MS_ADPCM_PREAMBLE_SIZE 6
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66
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67 #define LE_16(x) ((int16_t)AV_RL16(x))
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68
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69 // clamp a number between 0 and 88
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70 #define CLAMP_0_TO_88(x) x = av_clip(x, 0, 88);
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71 // clamp a number within a signed 16-bit range
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72 #define CLAMP_S16(x) x = av_clip_int16(x);
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73 // clamp a number above 16
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74 #define CLAMP_ABOVE_16(x) if (x < 16) x = 16;
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75 // sign extend a 4-bit value
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76 #define SE_4BIT(x) if (x & 0x8) x -= 0x10;
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77
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78 static int preinit(sh_audio_t *sh_audio)
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79 {
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80 sh_audio->audio_out_minsize = sh_audio->wf->nBlockAlign * 4;
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81 sh_audio->ds->ss_div =
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82 (sh_audio->wf->nBlockAlign - MS_ADPCM_PREAMBLE_SIZE) * 2;
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83 sh_audio->audio_in_minsize =
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84 sh_audio->ds->ss_mul = sh_audio->wf->nBlockAlign;
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85 return 1;
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86 }
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87
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88 static int init(sh_audio_t *sh_audio)
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89 {
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90 sh_audio->channels=sh_audio->wf->nChannels;
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91 sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
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92 sh_audio->i_bps = sh_audio->wf->nBlockAlign *
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93 (sh_audio->channels*sh_audio->samplerate) / sh_audio->ds->ss_div;
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94 sh_audio->samplesize=2;
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95
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96 return 1;
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97 }
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98
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99 static void uninit(sh_audio_t *sh_audio)
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100 {
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101 }
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102
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103 static int control(sh_audio_t *sh_audio,int cmd,void* arg, ...)
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104 {
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105 if(cmd==ADCTRL_SKIP_FRAME){
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106 demux_read_data(sh_audio->ds, sh_audio->a_in_buffer,sh_audio->ds->ss_mul);
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107 return CONTROL_TRUE;
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108 }
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109 return CONTROL_UNKNOWN;
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110 }
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111
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112 static inline int check_coeff(uint8_t c) {
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113 if (c > 6) {
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114 mp_msg(MSGT_DECAUDIO, MSGL_WARN,
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115 "MS ADPCM: coefficient (%d) out of range (should be [0..6])\n",
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116 c);
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117 c = 6;
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118 }
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119 return c;
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120 }
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121
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122 static int ms_adpcm_decode_block(unsigned short *output, unsigned char *input,
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123 int channels, int block_size)
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124 {
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125 int current_channel = 0;
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126 int coeff_idx;
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127 int idelta[2];
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128 int sample1[2];
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129 int sample2[2];
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130 int coeff1[2];
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131 int coeff2[2];
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132 int stream_ptr = 0;
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133 int out_ptr = 0;
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134 int upper_nibble = 1;
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135 int nibble;
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136 int snibble; // signed nibble
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137 int predictor;
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138
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139 if (channels != 1) channels = 2;
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140 if (block_size < 7 * channels)
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141 return -1;
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142
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143 // fetch the header information, in stereo if both channels are present
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144 coeff_idx = check_coeff(input[stream_ptr]);
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145 coeff1[0] = ms_adapt_coeff1[coeff_idx];
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146 coeff2[0] = ms_adapt_coeff2[coeff_idx];
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147 stream_ptr++;
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148 if (channels == 2)
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149 {
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150 coeff_idx = check_coeff(input[stream_ptr]);
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151 coeff1[1] = ms_adapt_coeff1[coeff_idx];
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152 coeff2[1] = ms_adapt_coeff2[coeff_idx];
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153 stream_ptr++;
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154 }
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155
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156 idelta[0] = LE_16(&input[stream_ptr]);
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157 stream_ptr += 2;
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158 if (channels == 2)
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159 {
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160 idelta[1] = LE_16(&input[stream_ptr]);
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161 stream_ptr += 2;
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162 }
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163
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164 sample1[0] = LE_16(&input[stream_ptr]);
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165 stream_ptr += 2;
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166 if (channels == 2)
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167 {
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168 sample1[1] = LE_16(&input[stream_ptr]);
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169 stream_ptr += 2;
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170 }
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171
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172 sample2[0] = LE_16(&input[stream_ptr]);
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173 stream_ptr += 2;
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174 if (channels == 2)
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175 {
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176 sample2[1] = LE_16(&input[stream_ptr]);
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177 stream_ptr += 2;
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178 }
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179
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180 if (channels == 1)
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181 {
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182 output[out_ptr++] = sample2[0];
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183 output[out_ptr++] = sample1[0];
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184 } else {
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185 output[out_ptr++] = sample2[0];
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186 output[out_ptr++] = sample2[1];
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187 output[out_ptr++] = sample1[0];
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188 output[out_ptr++] = sample1[1];
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189 }
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190
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191 while (stream_ptr < block_size)
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192 {
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193 // get the next nibble
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194 if (upper_nibble)
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195 nibble = snibble = input[stream_ptr] >> 4;
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196 else
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197 nibble = snibble = input[stream_ptr++] & 0x0F;
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198 upper_nibble ^= 1;
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199 SE_4BIT(snibble);
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200
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201 // should this really be a division and not a shift?
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202 // coefficients were originally scaled by for, which might have
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203 // been an optimization for 8-bit CPUs _if_ a shift is correct
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204 predictor = (
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205 ((sample1[current_channel] * coeff1[current_channel]) +
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206 (sample2[current_channel] * coeff2[current_channel])) / 64) +
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207 (snibble * idelta[current_channel]);
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208 CLAMP_S16(predictor);
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209 sample2[current_channel] = sample1[current_channel];
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210 sample1[current_channel] = predictor;
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211 output[out_ptr++] = predictor;
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212
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213 // compute the next adaptive scale factor (a.k.a. the variable idelta)
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214 idelta[current_channel] =
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215 (ms_adapt_table[nibble] * idelta[current_channel]) / 256;
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216 CLAMP_ABOVE_16(idelta[current_channel]);
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217
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218 // toggle the channel
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219 current_channel ^= channels - 1;
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220 }
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221
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222 return (block_size - (MS_ADPCM_PREAMBLE_SIZE * channels)) * 2;
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223 }
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224
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225 static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
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226 {
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227 int res;
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228 if (demux_read_data(sh_audio->ds, sh_audio->a_in_buffer,
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229 sh_audio->ds->ss_mul) !=
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230 sh_audio->ds->ss_mul)
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231 return -1; /* EOF */
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232
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233 res = ms_adpcm_decode_block(
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234 (unsigned short*)buf, sh_audio->a_in_buffer,
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235 sh_audio->wf->nChannels, sh_audio->wf->nBlockAlign);
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236 return res < 0 ? res : 2 * res;
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237 }