annotate libmpcodecs/ad_msadpcm.c @ 25471:5075d5ff1da8

Get rid of URLProtocol mess (especially problematic since it made use of a non-constant global variable) and use ByteIOContext directly.
author reimar
date Sat, 22 Dec 2007 16:22:54 +0000
parents fa99b3d31d13
children b21e1506e50b
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1 /*
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2 MS ADPCM Decoder for MPlayer
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3 by Mike Melanson
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4
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5 This file is responsible for decoding Microsoft ADPCM data.
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6 Details about the data format can be found here:
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7 http://www.pcisys.net/~melanson/codecs/
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8 */
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9
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10 #include <stdio.h>
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11 #include <stdlib.h>
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12 #include <unistd.h>
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13
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14 #include "config.h"
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15 #include "libavutil/common.h"
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16 #include "mpbswap.h"
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17 #include "ad_internal.h"
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18
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19 static ad_info_t info =
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20 {
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21 "MS ADPCM audio decoder",
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22 "msadpcm",
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23 "Nick Kurshev",
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24 "Mike Melanson",
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25 ""
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26 };
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27
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28 LIBAD_EXTERN(msadpcm)
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29
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30 static int ms_adapt_table[] =
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31 {
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32 230, 230, 230, 230, 307, 409, 512, 614,
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33 768, 614, 512, 409, 307, 230, 230, 230
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34 };
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35
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36 static int ms_adapt_coeff1[] =
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37 {
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38 256, 512, 0, 192, 240, 460, 392
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39 };
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40
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41 static int ms_adapt_coeff2[] =
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42 {
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43 0, -256, 0, 64, 0, -208, -232
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44 };
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45
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46 #define MS_ADPCM_PREAMBLE_SIZE 6
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47
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48 #define LE_16(x) ((x)[0]+(256*((x)[1])))
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49 //#define LE_16(x) (le2me_16((x)[1]+(256*((x)[0]))))
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50 //#define LE_16(x) (le2me_16(*(unsigned short *)(x)))
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51 //#define LE_32(x) (le2me_32(*(unsigned int *)(x)))
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52
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53 // useful macros
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54 // clamp a number between 0 and 88
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55 #define CLAMP_0_TO_88(x) if (x < 0) x = 0; else if (x > 88) x = 88;
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56 // clamp a number within a signed 16-bit range
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57 #define CLAMP_S16(x) if (x < -32768) x = -32768; \
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58 else if (x > 32767) x = 32767;
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59 // clamp a number above 16
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60 #define CLAMP_ABOVE_16(x) if (x < 16) x = 16;
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61 // sign extend a 16-bit value
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62 #define SE_16BIT(x) if (x & 0x8000) x -= 0x10000;
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63 // sign extend a 4-bit value
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64 #define SE_4BIT(x) if (x & 0x8) x -= 0x10;
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65
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66 static int preinit(sh_audio_t *sh_audio)
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67 {
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68 sh_audio->audio_out_minsize = sh_audio->wf->nBlockAlign * 4;
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69 sh_audio->ds->ss_div =
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70 (sh_audio->wf->nBlockAlign - MS_ADPCM_PREAMBLE_SIZE) * 2;
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71 sh_audio->audio_in_minsize =
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72 sh_audio->ds->ss_mul = sh_audio->wf->nBlockAlign;
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73 return 1;
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74 }
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75
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76 static int init(sh_audio_t *sh_audio)
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77 {
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78 sh_audio->channels=sh_audio->wf->nChannels;
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79 sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
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80 sh_audio->i_bps = sh_audio->wf->nBlockAlign *
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81 (sh_audio->channels*sh_audio->samplerate) / sh_audio->ds->ss_div;
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82 sh_audio->samplesize=2;
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83
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84 return 1;
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85 }
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86
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87 static void uninit(sh_audio_t *sh_audio)
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88 {
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89 }
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90
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91 static int control(sh_audio_t *sh_audio,int cmd,void* arg, ...)
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92 {
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93 if(cmd==ADCTRL_SKIP_FRAME){
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94 demux_read_data(sh_audio->ds, sh_audio->a_in_buffer,sh_audio->ds->ss_mul);
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95 return CONTROL_TRUE;
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96 }
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97 return CONTROL_UNKNOWN;
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98 }
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99
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100 static int ms_adpcm_decode_block(unsigned short *output, unsigned char *input,
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101 int channels, int block_size)
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102 {
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103 int current_channel = 0;
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104 int idelta[2];
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105 int sample1[2];
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106 int sample2[2];
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107 int coeff1[2];
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108 int coeff2[2];
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109 int stream_ptr = 0;
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110 int out_ptr = 0;
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111 int upper_nibble = 1;
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112 int nibble;
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113 int snibble; // signed nibble
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114 int predictor;
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115
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116 // fetch the header information, in stereo if both channels are present
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117 if (input[stream_ptr] > 6)
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118 mp_msg(MSGT_DECAUDIO, MSGL_WARN,
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119 "MS ADPCM: coefficient (%d) out of range (should be [0..6])\n",
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120 input[stream_ptr]);
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121 coeff1[0] = ms_adapt_coeff1[input[stream_ptr]];
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122 coeff2[0] = ms_adapt_coeff2[input[stream_ptr]];
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123 stream_ptr++;
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124 if (channels == 2)
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125 {
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126 if (input[stream_ptr] > 6)
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127 mp_msg(MSGT_DECAUDIO, MSGL_WARN,
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128 "MS ADPCM: coefficient (%d) out of range (should be [0..6])\n",
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129 input[stream_ptr]);
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130 coeff1[1] = ms_adapt_coeff1[input[stream_ptr]];
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131 coeff2[1] = ms_adapt_coeff2[input[stream_ptr]];
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132 stream_ptr++;
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133 }
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134
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135 idelta[0] = LE_16(&input[stream_ptr]);
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136 stream_ptr += 2;
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137 SE_16BIT(idelta[0]);
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138 if (channels == 2)
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139 {
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140 idelta[1] = LE_16(&input[stream_ptr]);
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141 stream_ptr += 2;
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142 SE_16BIT(idelta[1]);
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143 }
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144
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145 sample1[0] = LE_16(&input[stream_ptr]);
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146 stream_ptr += 2;
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147 SE_16BIT(sample1[0]);
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148 if (channels == 2)
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149 {
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150 sample1[1] = LE_16(&input[stream_ptr]);
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151 stream_ptr += 2;
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152 SE_16BIT(sample1[1]);
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153 }
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154
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155 sample2[0] = LE_16(&input[stream_ptr]);
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156 stream_ptr += 2;
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157 SE_16BIT(sample2[0]);
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158 if (channels == 2)
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159 {
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160 sample2[1] = LE_16(&input[stream_ptr]);
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161 stream_ptr += 2;
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162 SE_16BIT(sample2[1]);
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163 }
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164
6609
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165 if (channels == 1)
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166 {
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167 output[out_ptr++] = sample2[0];
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168 output[out_ptr++] = sample1[0];
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169 } else {
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170 output[out_ptr++] = sample2[0];
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171 output[out_ptr++] = sample2[1];
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172 output[out_ptr++] = sample1[0];
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173 output[out_ptr++] = sample1[1];
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174 }
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175
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176 while (stream_ptr < block_size)
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177 {
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178 // get the next nibble
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179 if (upper_nibble)
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180 nibble = snibble = input[stream_ptr] >> 4;
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181 else
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182 nibble = snibble = input[stream_ptr++] & 0x0F;
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183 upper_nibble ^= 1;
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184 SE_4BIT(snibble);
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185
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186 predictor = (
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187 ((sample1[current_channel] * coeff1[current_channel]) +
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188 (sample2[current_channel] * coeff2[current_channel])) / 256) +
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189 (snibble * idelta[current_channel]);
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190 CLAMP_S16(predictor);
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191 sample2[current_channel] = sample1[current_channel];
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192 sample1[current_channel] = predictor;
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193 output[out_ptr++] = predictor;
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194
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195 // compute the next adaptive scale factor (a.k.a. the variable idelta)
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196 idelta[current_channel] =
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197 (ms_adapt_table[nibble] * idelta[current_channel]) / 256;
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198 CLAMP_ABOVE_16(idelta[current_channel]);
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199
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200 // toggle the channel
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201 current_channel ^= channels - 1;
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202 }
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203
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204 return (block_size - (MS_ADPCM_PREAMBLE_SIZE * channels)) * 2;
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205 }
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206
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207 static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
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208 {
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209 if (demux_read_data(sh_audio->ds, sh_audio->a_in_buffer,
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210 sh_audio->ds->ss_mul) !=
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211 sh_audio->ds->ss_mul)
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212 return -1; /* EOF */
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213
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214 return 2 * ms_adpcm_decode_block(
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215 (unsigned short*)buf, sh_audio->a_in_buffer,
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216 sh_audio->wf->nChannels, sh_audio->wf->nBlockAlign);
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217 }