annotate libmpcodecs/ad_msadpcm.c @ 30537:7bebe34b6ad2

Move movie_aspect extern declaration to libmpcodecs/vd.h. This is the appropriate place for the declaration; the variable is declared in libmpcodecs/vd.c.
author diego
date Sun, 14 Feb 2010 14:47:22 +0000
parents cc27da5d7286
children a93891202051
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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
bbb6ebec87a0 Add missing license headers to all files in the libmpcodecs directory.
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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 "libavutil/common.h"
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33 #include "libavutil/intreadwrite.h"
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34 #include "mpbswap.h"
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35 #include "ad_internal.h"
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36
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37 static const ad_info_t info =
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38 {
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39 "MS ADPCM audio decoder",
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40 "msadpcm",
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41 "Nick Kurshev",
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42 "Mike Melanson",
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43 ""
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44 };
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45
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46 LIBAD_EXTERN(msadpcm)
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47
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48 static const int ms_adapt_table[] =
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49 {
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50 230, 230, 230, 230, 307, 409, 512, 614,
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51 768, 614, 512, 409, 307, 230, 230, 230
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52 };
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53
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54 static const uint8_t ms_adapt_coeff1[] =
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55 {
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56 64, 128, 0, 48, 60, 115, 98
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57 };
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58
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59 static const int8_t ms_adapt_coeff2[] =
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60 {
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61 0, -64, 0, 16, 0, -52, -58
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62 };
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63
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64 #define MS_ADPCM_PREAMBLE_SIZE 6
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65
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66 #define LE_16(x) ((int16_t)AV_RL16(x))
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67
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68 // clamp a number between 0 and 88
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69 #define CLAMP_0_TO_88(x) x = av_clip(x, 0, 88);
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70 // clamp a number within a signed 16-bit range
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71 #define CLAMP_S16(x) x = av_clip_int16(x);
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72 // clamp a number above 16
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73 #define CLAMP_ABOVE_16(x) if (x < 16) x = 16;
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74 // sign extend a 4-bit value
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75 #define SE_4BIT(x) if (x & 0x8) x -= 0x10;
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76
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77 static int preinit(sh_audio_t *sh_audio)
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78 {
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79 sh_audio->audio_out_minsize = sh_audio->wf->nBlockAlign * 4;
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80 sh_audio->ds->ss_div =
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81 (sh_audio->wf->nBlockAlign - MS_ADPCM_PREAMBLE_SIZE) * 2;
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82 sh_audio->audio_in_minsize =
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83 sh_audio->ds->ss_mul = sh_audio->wf->nBlockAlign;
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84 return 1;
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85 }
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86
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87 static int init(sh_audio_t *sh_audio)
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88 {
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89 sh_audio->channels=sh_audio->wf->nChannels;
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90 sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
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91 sh_audio->i_bps = sh_audio->wf->nBlockAlign *
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92 (sh_audio->channels*sh_audio->samplerate) / sh_audio->ds->ss_div;
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93 sh_audio->samplesize=2;
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94
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95 return 1;
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96 }
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97
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98 static void uninit(sh_audio_t *sh_audio)
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99 {
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100 }
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101
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102 static int control(sh_audio_t *sh_audio,int cmd,void* arg, ...)
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103 {
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104 if(cmd==ADCTRL_SKIP_FRAME){
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105 demux_read_data(sh_audio->ds, sh_audio->a_in_buffer,sh_audio->ds->ss_mul);
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106 return CONTROL_TRUE;
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107 }
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108 return CONTROL_UNKNOWN;
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109 }
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110
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111 static inline int check_coeff(uint8_t c) {
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112 if (c > 6) {
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113 mp_msg(MSGT_DECAUDIO, MSGL_WARN,
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114 "MS ADPCM: coefficient (%d) out of range (should be [0..6])\n",
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115 c);
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116 c = 6;
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117 }
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118 return c;
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119 }
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120
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121 static int ms_adpcm_decode_block(unsigned short *output, unsigned char *input,
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122 int channels, int block_size)
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123 {
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124 int current_channel = 0;
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125 int coeff_idx;
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126 int idelta[2];
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127 int sample1[2];
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128 int sample2[2];
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129 int coeff1[2];
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130 int coeff2[2];
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131 int stream_ptr = 0;
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132 int out_ptr = 0;
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133 int upper_nibble = 1;
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134 int nibble;
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135 int snibble; // signed nibble
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136 int predictor;
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137
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138 if (channels != 1) channels = 2;
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139 if (block_size < 7 * channels)
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140 return -1;
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141
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142 // fetch the header information, in stereo if both channels are present
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143 coeff_idx = check_coeff(input[stream_ptr]);
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144 coeff1[0] = ms_adapt_coeff1[coeff_idx];
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145 coeff2[0] = ms_adapt_coeff2[coeff_idx];
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146 stream_ptr++;
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147 if (channels == 2)
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148 {
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149 coeff_idx = check_coeff(input[stream_ptr]);
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150 coeff1[1] = ms_adapt_coeff1[coeff_idx];
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151 coeff2[1] = ms_adapt_coeff2[coeff_idx];
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152 stream_ptr++;
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153 }
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154
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155 idelta[0] = LE_16(&input[stream_ptr]);
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156 stream_ptr += 2;
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157 if (channels == 2)
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158 {
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159 idelta[1] = LE_16(&input[stream_ptr]);
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160 stream_ptr += 2;
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161 }
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162
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163 sample1[0] = LE_16(&input[stream_ptr]);
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164 stream_ptr += 2;
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165 if (channels == 2)
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166 {
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167 sample1[1] = LE_16(&input[stream_ptr]);
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168 stream_ptr += 2;
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169 }
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170
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171 sample2[0] = LE_16(&input[stream_ptr]);
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172 stream_ptr += 2;
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173 if (channels == 2)
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174 {
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175 sample2[1] = LE_16(&input[stream_ptr]);
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176 stream_ptr += 2;
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177 }
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178
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179 if (channels == 1)
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180 {
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181 output[out_ptr++] = sample2[0];
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182 output[out_ptr++] = sample1[0];
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183 } else {
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184 output[out_ptr++] = sample2[0];
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185 output[out_ptr++] = sample2[1];
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186 output[out_ptr++] = sample1[0];
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187 output[out_ptr++] = sample1[1];
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188 }
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189
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190 while (stream_ptr < block_size)
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191 {
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192 // get the next nibble
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193 if (upper_nibble)
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194 nibble = snibble = input[stream_ptr] >> 4;
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195 else
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196 nibble = snibble = input[stream_ptr++] & 0x0F;
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197 upper_nibble ^= 1;
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198 SE_4BIT(snibble);
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199
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200 // should this really be a division and not a shift?
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201 // coefficients were originally scaled by for, which might have
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202 // been an optimization for 8-bit CPUs _if_ a shift is correct
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203 predictor = (
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204 ((sample1[current_channel] * coeff1[current_channel]) +
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205 (sample2[current_channel] * coeff2[current_channel])) / 64) +
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206 (snibble * idelta[current_channel]);
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207 CLAMP_S16(predictor);
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208 sample2[current_channel] = sample1[current_channel];
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209 sample1[current_channel] = predictor;
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210 output[out_ptr++] = predictor;
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211
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212 // compute the next adaptive scale factor (a.k.a. the variable idelta)
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213 idelta[current_channel] =
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214 (ms_adapt_table[nibble] * idelta[current_channel]) / 256;
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215 CLAMP_ABOVE_16(idelta[current_channel]);
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216
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217 // toggle the channel
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218 current_channel ^= channels - 1;
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219 }
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220
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221 return (block_size - (MS_ADPCM_PREAMBLE_SIZE * channels)) * 2;
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222 }
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223
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224 static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
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225 {
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226 int res;
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227 if (demux_read_data(sh_audio->ds, sh_audio->a_in_buffer,
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228 sh_audio->ds->ss_mul) !=
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229 sh_audio->ds->ss_mul)
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230 return -1; /* EOF */
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231
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232 res = ms_adpcm_decode_block(
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233 (unsigned short*)buf, sh_audio->a_in_buffer,
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234 sh_audio->wf->nChannels, sh_audio->wf->nBlockAlign);
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235 return res < 0 ? res : 2 * res;
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236 }