Mercurial > SDL_sound_CoreAudio
annotate decoders/wav.c @ 202:ace10a8e632b
set -e and some more output.
author | Ryan C. Gordon <icculus@icculus.org> |
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date | Fri, 04 Jan 2002 06:51:25 +0000 |
parents | 6cd07211a235 |
children | c9772a9f5271 |
rev | line source |
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17 | 1 /* |
2 * SDL_sound -- An abstract sound format decoding API. | |
3 * Copyright (C) 2001 Ryan C. Gordon. | |
4 * | |
5 * This library is free software; you can redistribute it and/or | |
6 * modify it under the terms of the GNU Lesser General Public | |
7 * License as published by the Free Software Foundation; either | |
8 * version 2.1 of the License, or (at your option) any later version. | |
9 * | |
10 * This library is distributed in the hope that it will be useful, | |
11 * but WITHOUT ANY WARRANTY; without even the implied warranty of | |
12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | |
13 * Lesser General Public License for more details. | |
14 * | |
15 * You should have received a copy of the GNU Lesser General Public | |
16 * License along with this library; if not, write to the Free Software | |
17 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA | |
18 */ | |
19 | |
20 /* | |
21 * WAV decoder for SDL_sound. | |
22 * | |
23 * This driver handles Microsoft .WAVs, in as many of the thousands of | |
24 * variations as we can. | |
25 * | |
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26 * Please see the file COPYING in the source's root directory. |
17 | 27 * |
28 * This file written by Ryan C. Gordon. (icculus@clutteredmind.org) | |
29 */ | |
30 | |
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31 #if HAVE_CONFIG_H |
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32 # include <config.h> |
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33 #endif |
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34 |
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35 #ifdef SOUND_SUPPORTS_WAV |
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36 |
17 | 37 #include <stdio.h> |
38 #include <stdlib.h> | |
39 #include <string.h> | |
40 #include <assert.h> | |
41 | |
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42 #include "SDL_sound.h" |
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43 |
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44 #define __SDL_SOUND_INTERNAL__ |
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45 #include "SDL_sound_internal.h" |
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46 |
47
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47 static int WAV_init(void); |
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48 static void WAV_quit(void); |
17 | 49 static int WAV_open(Sound_Sample *sample, const char *ext); |
50 static void WAV_close(Sound_Sample *sample); | |
51 static Uint32 WAV_read(Sound_Sample *sample); | |
52 | |
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53 static const char *extensions_wav[] = { "WAV", NULL }; |
17 | 54 const Sound_DecoderFunctions __Sound_DecoderFunctions_WAV = |
55 { | |
56 { | |
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57 extensions_wav, |
17 | 58 "Microsoft WAVE audio format", |
59 "Ryan C. Gordon <icculus@clutteredmind.org>", | |
60 "http://www.icculus.org/SDL_sound/" | |
61 }, | |
62 | |
47
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63 WAV_init, /* init() method */ |
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64 WAV_quit, /* quit() method */ |
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65 WAV_open, /* open() method */ |
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66 WAV_close, /* close() method */ |
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67 WAV_read /* read() method */ |
17 | 68 }; |
69 | |
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70 |
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71 /* Better than SDL_ReadLE16, since you can detect i/o errors... */ |
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72 static inline int read_le16(SDL_RWops *rw, Uint16 *ui16) |
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73 { |
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74 int rc = SDL_RWread(rw, ui16, sizeof (Uint16), 1); |
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75 BAIL_IF_MACRO(rc != 1, ERR_IO_ERROR, 0); |
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76 *ui16 = SDL_SwapLE16(*ui16); |
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77 return(1); |
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78 } /* read_le16 */ |
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79 |
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80 |
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81 /* Better than SDL_ReadLE32, since you can detect i/o errors... */ |
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82 static inline int read_le32(SDL_RWops *rw, Uint32 *ui32) |
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83 { |
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84 int rc = SDL_RWread(rw, ui32, sizeof (Uint32), 1); |
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85 BAIL_IF_MACRO(rc != 1, ERR_IO_ERROR, 0); |
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86 *ui32 = SDL_SwapLE32(*ui32); |
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87 return(1); |
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88 } /* read_le32 */ |
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89 |
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90 |
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91 /* This is just cleaner on the caller's end... */ |
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92 static inline int read_uint8(SDL_RWops *rw, Uint8 *ui8) |
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93 { |
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94 int rc = SDL_RWread(rw, ui8, sizeof (Uint8), 1); |
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95 BAIL_IF_MACRO(rc != 1, ERR_IO_ERROR, 0); |
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96 return(1); |
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97 } /* read_uint8 */ |
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98 |
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99 |
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100 /* Chunk management code... */ |
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101 |
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102 #define riffID 0x46464952 /* "RIFF", in ascii. */ |
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103 #define waveID 0x45564157 /* "WAVE", in ascii. */ |
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104 #define factID 0x74636166 /* "fact", in ascii. */ |
124
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105 |
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106 |
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107 /***************************************************************************** |
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108 * The FORMAT chunk... * |
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109 *****************************************************************************/ |
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110 |
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111 #define fmtID 0x20746D66 /* "fmt ", in ascii. */ |
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112 |
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113 #define FMT_NORMAL 0x0001 /* Uncompressed waveform data. */ |
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114 #define FMT_ADPCM 0x0002 /* ADPCM compressed waveform data. */ |
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115 |
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116 typedef struct |
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117 { |
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118 Sint16 iCoef1; |
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119 Sint16 iCoef2; |
124
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120 } ADPCMCOEFSET; |
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121 |
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122 typedef struct |
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123 { |
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124 Uint8 bPredictor; |
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125 Uint16 iDelta; |
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126 Sint16 iSamp1; |
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127 Sint16 iSamp2; |
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128 } ADPCMBLOCKHEADER; |
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129 |
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130 typedef struct S_WAV_FMT_T |
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131 { |
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132 Uint32 chunkID; |
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133 Sint32 chunkSize; |
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134 Sint16 wFormatTag; |
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135 Uint16 wChannels; |
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136 Uint32 dwSamplesPerSec; |
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137 Uint32 dwAvgBytesPerSec; |
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138 Uint16 wBlockAlign; |
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139 Uint16 wBitsPerSample; |
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140 |
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141 Uint32 sample_frame_size; |
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142 |
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143 void (*free)(struct S_WAV_FMT_T *fmt); |
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144 Uint32 (*read_sample)(Sound_Sample *sample); |
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145 |
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146 union |
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147 { |
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148 struct |
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149 { |
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150 Uint16 cbSize; |
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151 Uint16 wSamplesPerBlock; |
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152 Uint16 wNumCoef; |
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153 ADPCMCOEFSET *aCoef; |
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154 ADPCMBLOCKHEADER *blockheaders; |
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155 Uint32 samples_left_in_block; |
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156 int nibble_state; |
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157 Sint8 nibble; |
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158 } adpcm; |
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159 |
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160 /* put other format-specific data here... */ |
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161 } fmt; |
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162 } fmt_t; |
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163 |
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164 |
17 | 165 /* |
166 * Read in a fmt_t from disk. This makes this process safe regardless of | |
167 * the processor's byte order or how the fmt_t structure is packed. | |
124
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168 * Note that the union "fmt" is not read in here; that is handled as |
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169 * needed in the read_fmt_* functions. |
17 | 170 */ |
171 static int read_fmt_chunk(SDL_RWops *rw, fmt_t *fmt) | |
172 { | |
173 /* skip reading the chunk ID, since it was already read at this point... */ | |
174 fmt->chunkID = fmtID; | |
175 | |
183
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176 BAIL_IF_MACRO(!read_le32(rw, &fmt->chunkSize), NULL, 0); |
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177 BAIL_IF_MACRO(!read_le16(rw, &fmt->wFormatTag), NULL, 0); |
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178 BAIL_IF_MACRO(!read_le16(rw, &fmt->wChannels), NULL, 0); |
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179 BAIL_IF_MACRO(!read_le32(rw, &fmt->dwSamplesPerSec), NULL, 0); |
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180 BAIL_IF_MACRO(!read_le32(rw, &fmt->dwAvgBytesPerSec), NULL, 0); |
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181 BAIL_IF_MACRO(!read_le16(rw, &fmt->wBlockAlign), NULL, 0); |
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182 BAIL_IF_MACRO(!read_le16(rw, &fmt->wBitsPerSample), NULL, 0); |
17 | 183 |
184 return(1); | |
185 } /* read_fmt_chunk */ | |
186 | |
187 | |
124
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188 |
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189 /***************************************************************************** |
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190 * The DATA chunk... * |
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191 *****************************************************************************/ |
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192 |
17 | 193 #define dataID 0x61746164 /* "data", in ascii. */ |
194 | |
195 typedef struct | |
196 { | |
197 Uint32 chunkID; | |
198 Sint32 chunkSize; | |
199 /* Then, (chunkSize) bytes of waveform data... */ | |
200 } data_t; | |
201 | |
202 | |
203 /* | |
124
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204 * Read in a data_t from disk. This makes this process safe regardless of |
17 | 205 * the processor's byte order or how the fmt_t structure is packed. |
206 */ | |
207 static int read_data_chunk(SDL_RWops *rw, data_t *data) | |
208 { | |
209 /* skip reading the chunk ID, since it was already read at this point... */ | |
210 data->chunkID = dataID; | |
183
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211 BAIL_IF_MACRO(!read_le32(rw, &data->chunkSize), NULL, 0); |
17 | 212 return(1); |
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213 } /* read_data_chunk */ |
17 | 214 |
215 | |
216 | |
217 | |
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218 /***************************************************************************** |
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219 * this is what we store in our internal->decoder_private field... * |
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220 *****************************************************************************/ |
17 | 221 |
124
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222 typedef struct |
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223 { |
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224 fmt_t *fmt; |
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225 Sint32 bytesLeft; |
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226 } wav_t; |
17 | 227 |
228 | |
229 | |
230 | |
124
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231 /***************************************************************************** |
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232 * Normal, uncompressed waveform handler... * |
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233 *****************************************************************************/ |
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234 |
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235 /* |
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236 * Sound_Decode() lands here for uncompressed WAVs... |
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237 */ |
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238 static Uint32 read_sample_fmt_normal(Sound_Sample *sample) |
17 | 239 { |
240 Uint32 retval; | |
241 Sound_SampleInternal *internal = (Sound_SampleInternal *) sample->opaque; | |
242 wav_t *w = (wav_t *) internal->decoder_private; | |
243 Uint32 max = (internal->buffer_size < (Uint32) w->bytesLeft) ? | |
244 internal->buffer_size : (Uint32) w->bytesLeft; | |
245 | |
246 assert(max > 0); | |
247 | |
248 /* | |
249 * We don't actually do any decoding, so we read the wav data | |
250 * directly into the internal buffer... | |
251 */ | |
252 retval = SDL_RWread(internal->rw, internal->buffer, 1, max); | |
253 | |
254 w->bytesLeft -= retval; | |
255 | |
256 /* Make sure the read went smoothly... */ | |
257 if ((retval == 0) || (w->bytesLeft == 0)) | |
258 sample->flags |= SOUND_SAMPLEFLAG_EOF; | |
259 | |
260 else if (retval == -1) | |
261 sample->flags |= SOUND_SAMPLEFLAG_ERROR; | |
262 | |
263 /* (next call this EAGAIN may turn into an EOF or error.) */ | |
264 else if (retval < internal->buffer_size) | |
265 sample->flags |= SOUND_SAMPLEFLAG_EAGAIN; | |
266 | |
267 return(retval); | |
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268 } /* read_sample_fmt_normal */ |
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269 |
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270 |
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271 |
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272 static int read_fmt_normal(SDL_RWops *rw, fmt_t *fmt) |
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273 { |
178
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274 /* (don't need to read more from the RWops...) */ |
183
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275 fmt->free = NULL; |
124
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276 fmt->read_sample = read_sample_fmt_normal; |
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277 return(1); |
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278 } /* read_fmt_normal */ |
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279 |
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280 |
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281 |
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282 /***************************************************************************** |
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283 * ADPCM compression handler... * |
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284 *****************************************************************************/ |
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285 |
185
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286 #define FIXED_POINT_COEF_BASE 256 |
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287 #define FIXED_POINT_ADAPTION_BASE 256 |
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288 #define SMALLEST_ADPCM_DELTA 16 |
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289 |
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290 |
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291 static inline int read_adpcm_block_headers(Sound_Sample *sample) |
183
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292 { |
185
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293 Sound_SampleInternal *internal = (Sound_SampleInternal *) sample->opaque; |
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294 SDL_RWops *rw = internal->rw; |
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295 wav_t *w = (wav_t *) internal->decoder_private; |
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296 fmt_t *fmt = w->fmt; |
183
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297 ADPCMBLOCKHEADER *headers = fmt->fmt.adpcm.blockheaders; |
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298 int i; |
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299 int max = fmt->wChannels; |
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300 |
185
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301 if (w->bytesLeft < fmt->wBlockAlign) |
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302 { |
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303 sample->flags |= SOUND_SAMPLEFLAG_EOF; |
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304 return(0); |
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305 } /* if */ |
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306 |
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307 w->bytesLeft -= fmt->wBlockAlign; |
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308 |
183
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309 for (i = 0; i < max; i++) |
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310 BAIL_IF_MACRO(!read_uint8(rw, &headers[i].bPredictor), NULL, 0); |
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311 |
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312 for (i = 0; i < max; i++) |
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313 BAIL_IF_MACRO(!read_le16(rw, &headers[i].iDelta), NULL, 0); |
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314 |
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315 for (i = 0; i < max; i++) |
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316 BAIL_IF_MACRO(!read_le16(rw, &headers[i].iSamp1), NULL, 0); |
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317 |
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318 for (i = 0; i < max; i++) |
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319 BAIL_IF_MACRO(!read_le16(rw, &headers[i].iSamp2), NULL, 0); |
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320 |
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321 fmt->fmt.adpcm.samples_left_in_block = fmt->fmt.adpcm.wSamplesPerBlock; |
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322 fmt->fmt.adpcm.nibble_state = 0; |
183
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323 return(1); |
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324 } /* read_adpcm_block_headers */ |
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325 |
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326 |
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327 static inline void do_adpcm_nibble(Uint8 nib, |
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328 ADPCMBLOCKHEADER *header, |
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329 Sint32 lPredSamp) |
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330 { |
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331 static const Sint32 max_audioval = ((1<<(16-1))-1); |
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332 static const Sint32 min_audioval = -(1<<(16-1)); |
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333 static const Sint32 AdaptionTable[] = |
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334 { |
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335 230, 230, 230, 230, 307, 409, 512, 614, |
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336 768, 614, 512, 409, 307, 230, 230, 230 |
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337 }; |
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338 |
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339 Sint32 lNewSamp; |
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340 Sint32 delta; |
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341 |
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342 if (nib & 0x08) |
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343 lNewSamp = lPredSamp + (header->iDelta * (nib - 0x10)); |
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344 else |
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345 lNewSamp = lPredSamp + (header->iDelta * nib); |
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346 |
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347 /* clamp value... */ |
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348 if (lNewSamp < min_audioval) |
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349 lNewSamp = min_audioval; |
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350 else if (lNewSamp > max_audioval) |
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351 lNewSamp = max_audioval; |
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352 |
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353 delta = ((Sint32) header->iDelta * AdaptionTable[nib]) / |
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354 FIXED_POINT_ADAPTION_BASE; |
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355 |
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356 if (delta < SMALLEST_ADPCM_DELTA) |
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357 delta = SMALLEST_ADPCM_DELTA; |
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358 |
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359 header->iDelta = delta; |
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360 header->iSamp2 = header->iSamp1; |
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361 header->iSamp1 = lNewSamp; |
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362 } /* do_adpcm_nibble */ |
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363 |
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364 |
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365 static inline int decode_adpcm_sample_frame(Sound_Sample *sample) |
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366 { |
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367 Sound_SampleInternal *internal = (Sound_SampleInternal *) sample->opaque; |
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368 wav_t *w = (wav_t *) internal->decoder_private; |
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369 fmt_t *fmt = w->fmt; |
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370 ADPCMBLOCKHEADER *headers = fmt->fmt.adpcm.blockheaders; |
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371 SDL_RWops *rw = internal->rw; |
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372 int i; |
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373 int max = fmt->wChannels; |
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374 Sint32 delta; |
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375 Uint8 nib = fmt->fmt.adpcm.nibble; |
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376 |
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377 for (i = 0; i < max; i++) |
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378 { |
185
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379 Uint8 byte; |
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380 Sint16 iCoef1 = fmt->fmt.adpcm.aCoef[headers[i].bPredictor].iCoef1; |
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381 Sint16 iCoef2 = fmt->fmt.adpcm.aCoef[headers[i].bPredictor].iCoef2; |
183
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382 Sint32 lPredSamp = ((headers[i].iSamp1 * iCoef1) + |
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383 (headers[i].iSamp2 * iCoef2)) / |
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384 FIXED_POINT_COEF_BASE; |
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385 |
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386 if (fmt->fmt.adpcm.nibble_state == 0) |
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387 { |
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388 BAIL_IF_MACRO(!read_uint8(rw, &nib), NULL, 0); |
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389 fmt->fmt.adpcm.nibble_state = 1; |
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390 do_adpcm_nibble(nib >> 4, &headers[i], lPredSamp); |
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391 } /* if */ |
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392 else |
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393 { |
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394 fmt->fmt.adpcm.nibble_state = 0; |
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395 do_adpcm_nibble(nib & 0x0F, &headers[i], lPredSamp); |
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396 } /* else */ |
183
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397 } /* for */ |
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398 |
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399 fmt->fmt.adpcm.nibble = nib; |
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400 return(1); |
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401 } /* decode_adpcm_sample_frame */ |
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402 |
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403 |
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404 static inline void put_adpcm_sample_frame1(Uint8 *_buf, fmt_t *fmt) |
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405 { |
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406 Uint16 *buf = (Uint16 *) _buf; |
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407 ADPCMBLOCKHEADER *headers = fmt->fmt.adpcm.blockheaders; |
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408 int i; |
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409 for (i = 0; i < fmt->wChannels; i++) |
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410 *(buf++) = headers[i].iSamp1; |
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411 } /* put_adpcm_sample_frame1 */ |
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412 |
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413 |
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414 static inline void put_adpcm_sample_frame2(Uint8 *_buf, fmt_t *fmt) |
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415 { |
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416 Uint16 *buf = (Uint16 *) _buf; |
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417 ADPCMBLOCKHEADER *headers = fmt->fmt.adpcm.blockheaders; |
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418 int i; |
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419 for (i = 0; i < fmt->wChannels; i++) |
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420 *(buf++) = headers[i].iSamp2; |
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421 } /* put_adpcm_sample_frame2 */ |
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422 |
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423 |
185
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424 /* |
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425 * Sound_Decode() lands here for ADPCM-encoded WAVs... |
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426 */ |
124
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427 static Uint32 read_sample_fmt_adpcm(Sound_Sample *sample) |
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428 { |
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429 Sound_SampleInternal *internal = (Sound_SampleInternal *) sample->opaque; |
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430 wav_t *w = (wav_t *) internal->decoder_private; |
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431 fmt_t *fmt = w->fmt; |
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432 Uint32 bw = 0; |
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433 |
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434 while (bw < internal->buffer_size) |
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435 { |
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436 /* write ongoing sample frame before reading more data... */ |
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437 switch (fmt->fmt.adpcm.samples_left_in_block) |
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438 { |
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439 case 0: /* need to read a new block... */ |
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440 if (!read_adpcm_block_headers(sample)) |
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441 { |
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442 if ((sample->flags & SOUND_SAMPLEFLAG_EOF) == 0) |
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443 sample->flags |= SOUND_SAMPLEFLAG_ERROR; |
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444 return(bw); |
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445 } /* if */ |
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446 |
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447 /* only write first sample frame for now. */ |
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448 put_adpcm_sample_frame2(internal->buffer + bw, fmt); |
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449 fmt->fmt.adpcm.samples_left_in_block--; |
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450 bw += fmt->sample_frame_size; |
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451 break; |
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452 |
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453 case 1: /* output last sample frame of block... */ |
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454 put_adpcm_sample_frame1(internal->buffer + bw, fmt); |
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455 fmt->fmt.adpcm.samples_left_in_block--; |
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456 bw += fmt->sample_frame_size; |
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457 break; |
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458 |
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459 default: /* output latest sample frame and read a new one... */ |
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460 put_adpcm_sample_frame1(internal->buffer + bw, fmt); |
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461 fmt->fmt.adpcm.samples_left_in_block--; |
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462 bw += fmt->sample_frame_size; |
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463 |
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464 if (!decode_adpcm_sample_frame(sample)) |
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465 { |
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466 sample->flags |= SOUND_SAMPLEFLAG_ERROR; |
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467 return(bw); |
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468 } /* if */ |
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469 } /* switch */ |
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470 } /* while */ |
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471 |
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472 return(bw); |
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473 } /* read_sample_fmt_adpcm */ |
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474 |
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475 |
185
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476 /* |
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477 * Sound_FreeSample() lands here for ADPCM-encoded WAVs... |
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478 */ |
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479 static void free_fmt_adpcm(fmt_t *fmt) |
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480 { |
183
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481 if (fmt->fmt.adpcm.aCoef != NULL) |
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482 free(fmt->fmt.adpcm.aCoef); |
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483 |
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484 if (fmt->fmt.adpcm.blockheaders != NULL) |
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485 free(fmt->fmt.adpcm.blockheaders); |
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486 } /* free_fmt_adpcm */ |
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487 |
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488 |
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489 /* |
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490 * Read in a the adpcm-specific info from disk. This makes this process |
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491 * safe regardless of the processor's byte order or how the fmt_t |
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492 * structure is packed. |
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493 */ |
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494 static int read_fmt_adpcm(SDL_RWops *rw, fmt_t *fmt) |
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495 { |
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496 size_t i; |
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497 |
183
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498 memset(&fmt->fmt.adpcm, '\0', sizeof (fmt->fmt.adpcm)); |
124
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499 fmt->free = free_fmt_adpcm; |
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500 fmt->read_sample = read_sample_fmt_adpcm; |
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501 |
183
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502 BAIL_IF_MACRO(!read_le16(rw, &fmt->fmt.adpcm.cbSize), NULL, 0); |
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503 BAIL_IF_MACRO(!read_le16(rw, &fmt->fmt.adpcm.wSamplesPerBlock), NULL, 0); |
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504 BAIL_IF_MACRO(!read_le16(rw, &fmt->fmt.adpcm.wNumCoef), NULL, 0); |
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505 |
185
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506 /* fmt->free() is always called, so these malloc()s will be cleaned up. */ |
124
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507 |
183
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508 i = sizeof (ADPCMCOEFSET) * fmt->fmt.adpcm.wNumCoef; |
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509 fmt->fmt.adpcm.aCoef = (ADPCMCOEFSET *) malloc(i); |
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510 BAIL_IF_MACRO(fmt->fmt.adpcm.aCoef == NULL, ERR_OUT_OF_MEMORY, 0); |
124
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511 |
185
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512 for (i = 0; i < fmt->fmt.adpcm.wNumCoef; i++) |
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513 { |
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514 BAIL_IF_MACRO(!read_le16(rw, &fmt->fmt.adpcm.aCoef[i].iCoef1), NULL, 0); |
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515 BAIL_IF_MACRO(!read_le16(rw, &fmt->fmt.adpcm.aCoef[i].iCoef2), NULL, 0); |
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516 } /* for */ |
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517 |
183
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518 i = sizeof (ADPCMBLOCKHEADER) * fmt->wChannels; |
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519 fmt->fmt.adpcm.blockheaders = (ADPCMBLOCKHEADER *) malloc(i); |
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520 BAIL_IF_MACRO(fmt->fmt.adpcm.blockheaders == NULL, ERR_OUT_OF_MEMORY, 0); |
124
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521 |
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522 return(1); |
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523 } /* read_fmt_adpcm */ |
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|
524 |
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525 |
178
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|
526 |
124
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527 /***************************************************************************** |
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528 * Everything else... * |
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529 *****************************************************************************/ |
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|
530 |
178
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diff
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|
531 static int WAV_init(void) |
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|
532 { |
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diff
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|
533 return(1); /* always succeeds. */ |
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534 } /* WAV_init */ |
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|
535 |
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|
536 |
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149
diff
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|
537 static void WAV_quit(void) |
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diff
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|
538 { |
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diff
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|
539 /* it's a no-op. */ |
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149
diff
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|
540 } /* WAV_quit */ |
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541 |
124
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542 |
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diff
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|
543 static int read_fmt(SDL_RWops *rw, fmt_t *fmt) |
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544 { |
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diff
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545 /* if it's in this switch statement, we support the format. */ |
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546 switch (fmt->wFormatTag) |
e46e31fdecfd
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|
547 { |
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|
548 case FMT_NORMAL: |
178
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|
549 SNDDBG(("WAV: Appears to be uncompressed audio.\n")); |
124
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diff
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|
550 return(read_fmt_normal(rw, fmt)); |
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diff
changeset
|
551 |
e46e31fdecfd
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diff
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|
552 case FMT_ADPCM: |
178
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diff
changeset
|
553 SNDDBG(("WAV: Appears to be ADPCM compressed audio.\n")); |
124
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554 return(read_fmt_adpcm(rw, fmt)); |
178
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|
555 |
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|
556 /* add other types here. */ |
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|
557 |
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|
558 default: |
193
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559 SNDDBG(("WAV: Format 0x%X is unknown.\n", |
178
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560 (unsigned int) fmt->wFormatTag)); |
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|
561 Sound_SetError("WAV: Unsupported format"); |
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|
562 return(0); /* not supported whatsoever. */ |
124
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diff
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|
563 } /* switch */ |
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564 |
178
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565 assert(0); /* shouldn't hit this point. */ |
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|
566 return(0); |
124
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diff
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|
567 } /* read_fmt */ |
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diff
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|
568 |
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106
diff
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|
569 |
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diff
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|
570 /* |
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diff
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571 * Locate a specific chunk in the WAVE file by ID... |
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572 */ |
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573 static int find_chunk(SDL_RWops *rw, Uint32 id) |
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574 { |
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|
575 Sint32 siz = 0; |
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576 Uint32 _id = 0; |
185
6bb68b3bdcf1
ADPCM decoder seems to be complete, other fixes too...
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|
577 Uint32 pos = SDL_RWtell(rw); |
124
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|
578 |
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579 while (1) |
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580 { |
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581 BAIL_IF_MACRO(!read_le32(rw, &_id), NULL, 0); |
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582 if (_id == id) |
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583 return(1); |
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584 |
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585 /* skip ahead and see what next chunk is... */ |
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586 BAIL_IF_MACRO(!read_le32(rw, &siz), NULL, 0); |
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587 assert(siz >= 0); |
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588 pos += (sizeof (Uint32) * 2) + siz; |
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589 if (siz > 0) |
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590 BAIL_IF_MACRO(SDL_RWseek(rw, pos, SEEK_SET) != pos, NULL, 0); |
124
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591 } /* while */ |
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592 |
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593 return(0); /* shouldn't hit this, but just in case... */ |
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594 } /* find_chunk */ |
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595 |
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596 |
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597 static int WAV_open_internal(Sound_Sample *sample, const char *ext, fmt_t *fmt) |
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598 { |
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599 Sound_SampleInternal *internal = (Sound_SampleInternal *) sample->opaque; |
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600 SDL_RWops *rw = internal->rw; |
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601 data_t d; |
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602 wav_t *w; |
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603 Uint32 pos; |
124
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604 |
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605 BAIL_IF_MACRO(SDL_ReadLE32(rw) != riffID, "WAV: Not a RIFF file.", 0); |
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606 SDL_ReadLE32(rw); /* throw the length away; we get this info later. */ |
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607 BAIL_IF_MACRO(SDL_ReadLE32(rw) != waveID, "WAV: Not a WAVE file.", 0); |
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608 BAIL_IF_MACRO(!find_chunk(rw, fmtID), "WAV: No format chunk.", 0); |
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609 BAIL_IF_MACRO(!read_fmt_chunk(rw, fmt), "WAV: Can't read format chunk.", 0); |
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610 |
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611 sample->actual.channels = (Uint8) fmt->wChannels; |
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612 sample->actual.rate = fmt->dwSamplesPerSec; |
193
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613 if ((fmt->wBitsPerSample == 4) /*|| (fmt->wBitsPerSample == 0) */ ) |
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614 sample->actual.format = AUDIO_S16SYS; /* !!! FIXME ? */ |
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615 else if (fmt->wBitsPerSample == 8) |
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616 sample->actual.format = AUDIO_U8; |
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617 else if (fmt->wBitsPerSample == 16) |
124
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618 sample->actual.format = AUDIO_S16LSB; |
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619 else |
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620 { |
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621 SNDDBG(("WAV: %d bits per sample!?\n", (int) fmt->wBitsPerSample)); |
124
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622 BAIL_MACRO("WAV: Unsupported sample size.", 0); |
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623 } /* else */ |
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624 |
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625 BAIL_IF_MACRO(!read_fmt(rw, fmt), NULL, 0); |
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626 BAIL_IF_MACRO(!find_chunk(rw, dataID), "WAV: No data chunk.", 0); |
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627 BAIL_IF_MACRO(!read_data_chunk(rw, &d), "WAV: Can't read data chunk.", 0); |
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628 |
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629 w = (wav_t *) malloc(sizeof(wav_t)); |
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630 BAIL_IF_MACRO(w == NULL, ERR_OUT_OF_MEMORY, 0); |
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631 w->fmt = fmt; |
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632 w->bytesLeft = d.chunkSize; |
185
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633 |
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634 /* !!! FIXME: Move this to Sound_SampleInfo ? */ |
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635 fmt->sample_frame_size = ( ((sample->actual.format & 0xFF) / 8) * |
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636 sample->actual.channels ); |
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637 |
124
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638 internal->decoder_private = (void *) w; |
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639 |
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640 sample->flags = SOUND_SAMPLEFLAG_NONE; |
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641 |
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642 SNDDBG(("WAV: Accepting data stream.\n")); |
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643 return(1); /* we'll handle this data. */ |
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644 } /* WAV_open_internal */ |
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645 |
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646 |
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647 static int WAV_open(Sound_Sample *sample, const char *ext) |
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648 { |
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649 int rc; |
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650 |
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651 fmt_t *fmt = (fmt_t *) malloc(sizeof (fmt_t)); |
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652 BAIL_IF_MACRO(fmt == NULL, ERR_OUT_OF_MEMORY, 0); |
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653 memset(fmt, '\0', sizeof (fmt_t)); |
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654 |
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655 rc = WAV_open_internal(sample, ext, fmt); |
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656 if (!rc) |
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657 { |
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658 if (fmt->free != NULL) |
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659 fmt->free(fmt); |
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660 free(fmt); |
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661 } /* if */ |
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662 |
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663 return(rc); |
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664 } /* WAV_open */ |
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665 |
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|
666 |
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667 static void WAV_close(Sound_Sample *sample) |
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668 { |
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669 Sound_SampleInternal *internal = (Sound_SampleInternal *) sample->opaque; |
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670 wav_t *w = (wav_t *) internal->decoder_private; |
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671 |
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672 if (w->fmt->free != NULL) |
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673 w->fmt->free(w->fmt); |
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674 |
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675 free(w->fmt); |
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|
676 free(w); |
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677 } /* WAV_close */ |
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678 |
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|
679 |
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680 static Uint32 WAV_read(Sound_Sample *sample) |
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681 { |
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682 Sound_SampleInternal *internal = (Sound_SampleInternal *) sample->opaque; |
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683 wav_t *w = (wav_t *) internal->decoder_private; |
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684 return(w->fmt->read_sample(sample)); |
17 | 685 } /* WAV_read */ |
686 | |
64
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687 #endif /* SOUND_SUPPORTS_WAV */ |
17 | 688 |
689 /* end of wav.c ... */ | |
690 |