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428
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1 /* Play sound using the SGI audio library
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2 written by Simon Leinen <simon@lia.di.epfl.ch>
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3 Copyright (C) 1992 Free Software Foundation, Inc.
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4
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5 This file is part of XEmacs.
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6
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7 XEmacs is free software; you can redistribute it and/or modify it
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8 under the terms of the GNU General Public License as published by the
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9 Free Software Foundation; either version 2, or (at your option) any
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10 later version.
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11
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12 XEmacs is distributed in the hope that it will be useful, but WITHOUT
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13 ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or
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14 FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License
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15 for more details.
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16
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17 You should have received a copy of the GNU General Public License
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18 along with XEmacs; see the file COPYING. If not, write to
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19 the Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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20 Boston, MA 02111-1307, USA. */
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21
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22 /* Synched up with: Not in FSF. */
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23
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563
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24 /* This file Mule-ized by Ben Wing, 5-15-01. */
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25
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428
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26 #include <config.h>
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27 #include "lisp.h"
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28
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563
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29 #include "sound.h"
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30
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31 #include "sysfile.h"
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32 #include "sysproc.h" /* netinet/in.h for ntohl() etc. */
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428
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33
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34 /* Configuration options */
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35
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36 /* ability to parse Sun/NeXT (.au or .snd) audio file headers. The
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37 .snd format supports all sampling rates and sample widths that are
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38 commonly used, as well as stereo. It is also easy to parse. */
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39 #ifndef HAVE_SND_FILES
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40 #define HAVE_SND_FILES 1
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41 #endif
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42
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43 /* support for eight-but mu-law encoding. This is a useful compaction
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44 technique, and most sounds from the Sun universe are in this
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45 format. */
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46 #ifndef HAVE_MULAW_8
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47 #define HAVE_MULAW_8 1
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48 #endif
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49
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50 /* if your machine is very slow, you have to use a table lookup to
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51 convert mulaw samples to linear. This makes Emacs bigger so try to
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52 avoid it. */
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53 #ifndef USE_MULAW_DECODE_TABLE
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54 #define USE_MULAW_DECODE_TABLE 0
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55 #endif
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56
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57 /* support for linear encoding -- useful if you want better quality.
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58 This enables 8, 16 and 24 bit wide samples. */
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59 #ifndef HAVE_LINEAR
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60 #define HAVE_LINEAR 1
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61 #endif
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62
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63 /* support for 32 bit wide samples. If you notice the difference
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64 between 32 and 24 bit samples, you must have very good ears. Since
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65 the SGI audio library only supports 24 bit samples, each sample has
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66 to be shifted right by 8 bits anyway. So you should probably just
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67 convert all your 32 bit audio files to 24 bit. */
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68 #ifndef HAVE_LINEAR_32
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69 #define HAVE_LINEAR_32 0
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70 #endif
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71
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72 /* support for stereo sound. Imagine the cool applications of this:
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73 finally you don't just hear a beep -- you also know immediately
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74 *where* something went wrong! Unfortunately the programming
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75 interface only takes a single volume argument so far. */
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76 #ifndef HAVE_STEREO
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77 #define HAVE_STEREO 1
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78 #endif
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79
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80 /* the play routine can be interrupted between chunks, so we choose a
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81 small chunksize to keep the system responsive (2000 samples
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82 correspond to a quarter of a second for .au files. If you
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83 HAVE_STEREO, the chunksize should probably be even. */
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84 #define CHUNKSIZE 8000
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85
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86 /* the format assumed for header-less audio data. The following
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87 assumes ".au" format (8000 samples/sec mono 8-bit mulaw). */
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88 #define DEFAULT_SAMPLING_RATE 8000
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89 #define DEFAULT_CHANNEL_COUNT 1
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90 #define DEFAULT_FORMAT AFmulaw8
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91
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92 /* Data structures */
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93
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94 /* an AudioContext describes everything we want to know about how a
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95 particular sound snippet should be played. It is split into three
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96 parts (device, port and buffer) for implementation reasons. The
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97 device part corresponds to the state of the output device and must
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98 be reverted after playing the samples. The port part corresponds
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99 to an ALport; we want to allocate a minimal number of these since
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100 there are only four of them system-wide, but on the other hand we
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101 can't use the same port for mono and stereo. The buffer part
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102 corresponds to the sound data itself. */
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103
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104 typedef struct _AudioContextRec * AudioContext;
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105
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106 typedef struct
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107 {
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108 long device;
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109 int left_speaker_gain;
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110 int right_speaker_gain;
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111 long output_rate;
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112 }
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113 AudioDeviceRec, * AudioDevice;
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114
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115 /* supported sound data formats */
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116
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117 typedef enum
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118 {
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119 AFunknown,
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120 #if HAVE_MULAW_8
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121 AFmulaw8,
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122 #endif
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123 #if HAVE_LINEAR
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124 AFlinear8,
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125 AFlinear16,
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126 AFlinear24,
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127 #if HAVE_LINEAR_32
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128 AFlinear32,
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129 #endif
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130 #endif
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131 AFillegal
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132 }
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133 AudioFormat;
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134
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135 typedef struct
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136 {
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137 ALport port;
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138 AudioFormat format;
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139 unsigned nchan;
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140 unsigned queue_size;
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141 }
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142 AudioPortRec, * AudioPort;
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143
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144 typedef struct
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145 {
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146 void * data;
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147 unsigned long size;
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148 void (* write_chunk_function) (void *, void *, AudioContext);
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149 }
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150 AudioBufferRec, * AudioBuffer;
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151
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152 typedef struct _AudioContextRec
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153 {
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154 AudioDeviceRec device;
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155 AudioPortRec port;
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156 AudioBufferRec buffer;
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157 }
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158 AudioContextRec;
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159
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160 #define ac_device device.device
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161 #define ac_left_speaker_gain device.left_speaker_gain
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162 #define ac_right_speaker_gain device.right_speaker_gain
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163 #define ac_output_rate device.output_rate
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164 #define ac_port port.port
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165 #define ac_format port.format
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166 #define ac_nchan port.nchan
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167 #define ac_queue_size port.queue_size
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168 #define ac_data buffer.data
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169 #define ac_size buffer.size
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170 #define ac_write_chunk_function buffer.write_chunk_function
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171
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172 /* Forward declarations */
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173
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174 static Lisp_Object close_sound_file (Lisp_Object);
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563
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175 static AudioContext audio_initialize (UChar_Binary *, int, int);
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176 static void play_internal (UChar_Binary *, int, AudioContext);
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428
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177 static void drain_audio_port (AudioContext);
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178 static void write_mulaw_8_chunk (void *, void *, AudioContext);
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179 static void write_linear_chunk (void *, void *, AudioContext);
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180 static void write_linear_32_chunk (void *, void *, AudioContext);
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181 static Lisp_Object restore_audio_port (Lisp_Object);
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182 static AudioContext initialize_audio_port (AudioContext);
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183 static int open_audio_port (AudioContext, AudioContext);
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184 static void adjust_audio_volume (AudioDevice);
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185 static void get_current_volumes (AudioDevice);
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186 static int set_channels (ALconfig, unsigned);
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187 static int set_output_format (ALconfig, AudioFormat);
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188 static int parse_snd_header (void*, long, AudioContext);
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189
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190 /* are we looking at an NeXT/Sun audio header? */
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191 #define LOOKING_AT_SND_HEADER_P(address) \
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192 (!strncmp(".snd", (char *)(address), 4))
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193
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194 static Lisp_Object
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195 close_sound_file (Lisp_Object closure)
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196 {
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197 close (XINT (closure));
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198 return Qnil;
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199 }
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200
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201 void
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563
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202 play_sound_file (Extbyte *sound_file, int volume)
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428
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203 {
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204 int count = specpdl_depth ();
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205 int input_fd;
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563
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206 UChar_Binary buffer[CHUNKSIZE];
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428
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207 int bytes_read;
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208 AudioContext ac = (AudioContext) 0;
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209
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210 input_fd = open (sound_file, O_RDONLY);
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211 if (input_fd == -1)
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212 /* no error message -- this can't happen
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213 because Fplay_sound_file has checked the
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214 file for us. */
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215 return;
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216
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217 record_unwind_protect (close_sound_file, make_int (input_fd));
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218
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219 while ((bytes_read = read (input_fd, buffer, CHUNKSIZE)) > 0)
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220 {
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221 if (ac == (AudioContext) 0)
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222 {
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223 ac = audio_initialize (buffer, bytes_read, volume);
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224 if (ac == 0)
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225 return;
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226 }
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227 else
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228 {
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229 ac->ac_data = buffer;
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230 ac->ac_size = bytes_read;
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231 }
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232 play_internal (buffer, bytes_read, ac);
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233 }
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234 drain_audio_port (ac);
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235 unbind_to (count, Qnil);
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236 }
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237
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238 static long
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239 saved_device_state[] = {
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240 AL_OUTPUT_RATE, 0,
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241 AL_LEFT_SPEAKER_GAIN, 0,
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242 AL_RIGHT_SPEAKER_GAIN, 0,
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243 };
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244
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245 static Lisp_Object
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246 restore_audio_port (Lisp_Object closure)
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247 {
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248 Lisp_Object * contents = XVECTOR_DATA (closure);
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249 saved_device_state[1] = XINT (contents[0]);
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250 saved_device_state[3] = XINT (contents[1]);
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251 saved_device_state[5] = XINT (contents[2]);
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252 ALsetparams (AL_DEFAULT_DEVICE, saved_device_state, 6);
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253 return Qnil;
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254 }
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255
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256 void
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563
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257 play_sound_data (UChar_Binary *data, int length, int volume)
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428
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258 {
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259 int count = specpdl_depth ();
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260 AudioContext ac;
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261
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262 ac = audio_initialize (data, length, volume);
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263 if (ac == (AudioContext) 0)
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264 return;
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265 play_internal (data, length, ac);
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266 drain_audio_port (ac);
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267 unbind_to (count, Qnil);
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268 }
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269
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270 static AudioContext
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563
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271 audio_initialize (UChar_Binary *data, int length, int volume)
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428
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272 {
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273 Lisp_Object audio_port_state[3];
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274 static AudioContextRec desc;
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275 AudioContext ac;
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276
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277 desc.ac_right_speaker_gain
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278 = desc.ac_left_speaker_gain
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279 = volume * 256 / 100;
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280 desc.ac_device = AL_DEFAULT_DEVICE;
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281
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282 #if HAVE_SND_FILES
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283 if (LOOKING_AT_SND_HEADER_P (data))
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284 {
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285 if (parse_snd_header (data, length, & desc)==-1)
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563
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286 report_sound_error ("decoding .snd header", Qunbound);
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428
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287 }
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288 else
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289 #endif
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290 {
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291 desc.ac_data = data;
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292 desc.ac_size = length;
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293 desc.ac_output_rate = DEFAULT_SAMPLING_RATE;
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294 desc.ac_nchan = DEFAULT_CHANNEL_COUNT;
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295 desc.ac_format = DEFAULT_FORMAT;
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296 desc.ac_write_chunk_function = write_mulaw_8_chunk;
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297 }
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298
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299 /* Make sure that the audio port is reset to
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300 its initial characteristics after exit */
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301 ALgetparams (desc.ac_device, saved_device_state,
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302 sizeof (saved_device_state) / sizeof (long));
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303 audio_port_state[0] = make_int (saved_device_state[1]);
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304 audio_port_state[1] = make_int (saved_device_state[3]);
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305 audio_port_state[2] = make_int (saved_device_state[5]);
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306 record_unwind_protect (restore_audio_port,
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307 Fvector (3, &audio_port_state[0]));
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308
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309 ac = initialize_audio_port (& desc);
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310 desc = * ac;
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311 return ac;
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312 }
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313
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314 static void
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563
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315 play_internal (UChar_Binary *data, int length, AudioContext ac)
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428
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316 {
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563
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317 UChar_Binary * limit;
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428
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318 if (ac == (AudioContext) 0)
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319 return;
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320
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563
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321 data = (UChar_Binary *) ac->ac_data;
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428
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322 limit = data + ac->ac_size;
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323 while (data < limit)
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324 {
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563
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325 UChar_Binary * chunklimit = data + CHUNKSIZE;
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428
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326
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327 if (chunklimit > limit)
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328 chunklimit = limit;
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329
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330 QUIT;
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331
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332 (* ac->ac_write_chunk_function) (data, chunklimit, ac);
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333 data = chunklimit;
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334 }
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335 }
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336
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337 static void
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338 drain_audio_port (AudioContext ac)
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339 {
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340 while (ALgetfilled (ac->ac_port) > 0)
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341 sginap(1);
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342 }
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343
|
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344 /* Methods to write a "chunk" from a buffer containing audio data to
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345 an audio port. This may involve some conversion if the output
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346 device doesn't directly support the format the audio data is in. */
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347
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348 #if HAVE_MULAW_8
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349
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|
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350 #if USE_MULAW_DECODE_TABLE
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351 #include "libst.h"
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352 #else /* not USE_MULAW_DECODE_TABLE */
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353 static int
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354 st_ulaw_to_linear (int u)
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|
|
355 {
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442
|
356 static const short table[] = {0,132,396,924,1980,4092,8316,16764};
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|
428
|
357 int u1 = ~u;
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|
|
358 short exponent = (u1 >> 4) & 0x07;
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|
|
359 int mantissa = u1 & 0x0f;
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|
|
360 int unsigned_result = table[exponent]+(mantissa << (exponent+3));
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|
|
361 return u1 & 0x80 ? -unsigned_result : unsigned_result;
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|
|
362 }
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|
|
363 #endif /* not USE_MULAW_DECODE_TABLE */
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|
|
364
|
|
|
365 static void
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|
|
366 write_mulaw_8_chunk (void *buffer, void *chunklimit, AudioContext ac)
|
|
|
367 {
|
|
563
|
368 UChar_Binary * data = (UChar_Binary *) buffer;
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|
|
369 UChar_Binary * limit = (UChar_Binary *) chunklimit;
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|
428
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370 short * obuf, * bufp;
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|
371 long n_samples = limit - data;
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|
|
372
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|
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373 obuf = alloca_array (short, n_samples);
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|
374 bufp = &obuf[0];
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375
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|
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376 while (data < limit)
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|
377 *bufp++ = st_ulaw_to_linear (*data++);
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|
|
378 ALwritesamps (ac->ac_port, obuf, n_samples);
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|
|
379 }
|
|
|
380 #endif /* HAVE_MULAW_8 */
|
|
|
381
|
|
|
382 #if HAVE_LINEAR
|
|
|
383 static void
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|
|
384 write_linear_chunk (void *data, void *limit, AudioContext ac)
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|
|
385 {
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|
|
386 unsigned n_samples;
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|
|
387
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|
|
388 switch (ac->ac_format)
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|
|
389 {
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|
|
390 case AFlinear16: n_samples = (short *) limit - (short *) data; break;
|
|
563
|
391 case AFlinear8: n_samples = (Char_Binary *) limit - (Char_Binary *) data; break;
|
|
428
|
392 default: n_samples = (long *) limit - (long *) data; break;
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|
|
393 }
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|
|
394 ALwritesamps (ac->ac_port, data, (long) n_samples);
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|
|
395 }
|
|
|
396
|
|
|
397 #if HAVE_LINEAR_32
|
|
|
398 static void
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|
|
399 write_linear_32_chunk (void *buffer, void *chunklimit, AudioContext ac)
|
|
|
400 {
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|
|
401 long * data = (long *) buffer;
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|
|
402 long * limit = (long *) chunklimit;
|
|
|
403 long * obuf, * bufp;
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|
|
404 long n_samples = limit-data;
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|
|
405
|
|
|
406 obuf = alloca_array (long, n_samples);
|
|
|
407 bufp = &obuf[0];
|
|
|
408
|
|
|
409 while (data < limit)
|
|
|
410 *bufp++ = *data++ >> 8;
|
|
|
411 ALwritesamps (ac->ac_port, obuf, n_samples);
|
|
|
412 }
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|
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413 #endif /* HAVE_LINEAR_32 */
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414 #endif /* HAVE_LINEAR */
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415
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416 static AudioContext
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417 initialize_audio_port (AudioContext desc)
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418 {
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419 /* we can't use the same port for mono and stereo */
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420 static AudioContextRec mono_port_state
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421 = { { 0, 0, 0, 0 },
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422 { (ALport) 0, AFunknown, 1, 0 },
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423 { (void *) 0, (unsigned long) 0 } };
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424 #if HAVE_STEREO
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425 static AudioContextRec stereo_port_state
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426 = { { 0, 0, 0, 0 },
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427 { (ALport) 0, AFunknown, 2, 0 },
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428 { (void *) 0, (unsigned long) 0 } };
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|
|
429 static AudioContext return_ac;
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430
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431 switch (desc->ac_nchan)
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432 {
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433 case 1: return_ac = & mono_port_state; break;
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434 case 2: return_ac = & stereo_port_state; break;
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435 default: return (AudioContext) 0;
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436 }
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437 #else /* not HAVE_STEREO */
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438 static AudioContext return_ac = & mono_port_state;
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439 #endif /* not HAVE_STEREO */
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440
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441 return_ac->device = desc->device;
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442 return_ac->buffer = desc->buffer;
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443 return_ac->ac_format = desc->ac_format;
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444 return_ac->ac_queue_size = desc->ac_queue_size;
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445
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446 if (return_ac->ac_port==(ALport) 0)
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447 {
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|
448 if ((open_audio_port (return_ac, desc))==-1)
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449 {
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563
|
450 report_sound_error ("Open audio port", Qunbound);
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|
428
|
451 return (AudioContext) 0;
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452 }
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453 }
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454 else
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455 {
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456 ALconfig config = ALgetconfig (return_ac->ac_port);
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457 int changed = 0;
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|
458 long params[2];
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459
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|
|
460 params[0] = AL_OUTPUT_RATE;
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|
461 ALgetparams (return_ac->ac_device, params, 2);
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462 return_ac->ac_output_rate = params[1];
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463
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464 if (return_ac->ac_output_rate != desc->ac_output_rate)
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465 {
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|
466 return_ac->ac_output_rate = params[1] = desc->ac_output_rate;
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|
467 ALsetparams (return_ac->ac_device, params, 2);
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|
|
468 }
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|
469 if ((changed = set_output_format (config, return_ac->ac_format))==-1)
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|
470 return (AudioContext) 0;
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|
471 return_ac->ac_format = desc->ac_format;
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|
|
472 if (changed)
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|
473 ALsetconfig (return_ac->ac_port, config);
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|
|
474 }
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|
|
475 return_ac->ac_write_chunk_function = desc->ac_write_chunk_function;
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|
476 get_current_volumes (& return_ac->device);
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|
|
477 if (return_ac->ac_left_speaker_gain != desc->ac_left_speaker_gain
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|
478 || return_ac->ac_right_speaker_gain != desc->ac_right_speaker_gain)
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|
|
479 adjust_audio_volume (& desc->device);
|
|
|
480 return return_ac;
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|
|
481 }
|
|
|
482
|
|
|
483 static int
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|
|
484 open_audio_port (AudioContext return_ac, AudioContext desc)
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|
|
485 {
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|
|
486 ALconfig config = ALnewconfig();
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|
|
487 long params[2];
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|
|
488
|
|
|
489 adjust_audio_volume (& desc->device);
|
|
|
490 return_ac->ac_left_speaker_gain = desc->ac_left_speaker_gain;
|
|
|
491 return_ac->ac_right_speaker_gain = desc->ac_right_speaker_gain;
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|
|
492 params[0] = AL_OUTPUT_RATE;
|
|
|
493 params[1] = desc->ac_output_rate;
|
|
|
494 ALsetparams (desc->ac_device, params, 2);
|
|
|
495 return_ac->ac_output_rate = desc->ac_output_rate;
|
|
|
496 if (set_channels (config, desc->ac_nchan)==-1)
|
|
|
497 return -1;
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|
|
498 return_ac->ac_nchan = desc->ac_nchan;
|
|
|
499 if (set_output_format (config, desc->ac_format)==-1)
|
|
|
500 return -1;
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|
|
501 return_ac->ac_format = desc->ac_format;
|
|
|
502 ALsetqueuesize (config, (long) CHUNKSIZE);
|
|
|
503 return_ac->ac_port = ALopenport("XEmacs audio output", "w", config);
|
|
|
504 ALfreeconfig (config);
|
|
|
505 if (return_ac->ac_port==0)
|
|
|
506 {
|
|
563
|
507 report_sound_error ("Opening audio output port", Qunbound);
|
|
428
|
508 return -1;
|
|
|
509 }
|
|
|
510 return 0;
|
|
|
511 }
|
|
|
512
|
|
|
513 static int
|
|
|
514 set_channels (ALconfig config, unsigned int nchan)
|
|
|
515 {
|
|
|
516 switch (nchan)
|
|
|
517 {
|
|
|
518 case 1: ALsetchannels (config, AL_MONO); break;
|
|
|
519 #if HAVE_STEREO
|
|
|
520 case 2: ALsetchannels (config, AL_STEREO); break;
|
|
|
521 #endif /* HAVE_STEREO */
|
|
|
522 default:
|
|
563
|
523 report_sound_error ("Unsupported channel count",
|
|
|
524 make_int (nchan));
|
|
428
|
525 return -1;
|
|
|
526 }
|
|
|
527 return 0;
|
|
|
528 }
|
|
|
529
|
|
|
530 static int
|
|
|
531 set_output_format (ALconfig config, AudioFormat format)
|
|
|
532 {
|
|
|
533 long samplesize;
|
|
|
534 long old_samplesize;
|
|
|
535
|
|
|
536 switch (format)
|
|
|
537 {
|
|
|
538 #if HAVE_MULAW_8
|
|
|
539 case AFmulaw8:
|
|
|
540 #endif
|
|
|
541 #if HAVE_LINEAR
|
|
|
542 case AFlinear16:
|
|
|
543 #endif
|
|
|
544 #if HAVE_MULAW_8 || HAVE_LINEAR
|
|
|
545 samplesize = AL_SAMPLE_16;
|
|
|
546 break;
|
|
|
547 #endif
|
|
|
548 #if HAVE_LINEAR
|
|
|
549 case AFlinear8:
|
|
|
550 samplesize = AL_SAMPLE_8;
|
|
|
551 break;
|
|
|
552 case AFlinear24:
|
|
|
553 #if HAVE_LINEAR_32
|
|
|
554 case AFlinear32:
|
|
|
555 samplesize = AL_SAMPLE_24;
|
|
|
556 break;
|
|
|
557 #endif
|
|
|
558 #endif
|
|
|
559 default:
|
|
563
|
560 report_sound_error ("Unsupported audio format", make_int (format));
|
|
428
|
561 return -1;
|
|
|
562 }
|
|
|
563 old_samplesize = ALgetwidth (config);
|
|
|
564 if (old_samplesize==samplesize)
|
|
|
565 return 0;
|
|
|
566 ALsetwidth (config, samplesize);
|
|
|
567 return 1;
|
|
|
568 }
|
|
|
569
|
|
|
570 static void
|
|
|
571 adjust_audio_volume (AudioDevice device)
|
|
|
572 {
|
|
|
573 long params[4];
|
|
|
574 params[0] = AL_LEFT_SPEAKER_GAIN;
|
|
|
575 params[1] = device->left_speaker_gain;
|
|
|
576 params[2] = AL_RIGHT_SPEAKER_GAIN;
|
|
|
577 params[3] = device->right_speaker_gain;
|
|
|
578 ALsetparams (device->device, params, 4);
|
|
|
579 }
|
|
|
580
|
|
|
581 static void
|
|
|
582 get_current_volumes (AudioDevice device)
|
|
|
583 {
|
|
|
584 long params[4];
|
|
|
585 params[0] = AL_LEFT_SPEAKER_GAIN;
|
|
|
586 params[2] = AL_RIGHT_SPEAKER_GAIN;
|
|
|
587 ALgetparams (device->device, params, 4);
|
|
|
588 device->left_speaker_gain = params[1];
|
|
|
589 device->right_speaker_gain = params[3];
|
|
|
590 }
|
|
|
591
|
|
|
592 #if HAVE_SND_FILES
|
|
|
593
|
|
|
594 /* Parsing .snd (NeXT/Sun) headers */
|
|
|
595
|
|
|
596 typedef struct
|
|
|
597 {
|
|
|
598 int magic;
|
|
|
599 int dataLocation;
|
|
|
600 int dataSize;
|
|
|
601 int dataFormat;
|
|
|
602 int samplingRate;
|
|
|
603 int channelCount;
|
|
563
|
604 Char_Binary info[4];
|
|
428
|
605 }
|
|
|
606 SNDSoundStruct;
|
|
|
607 #define SOUND_TO_HOST_INT(x) ntohl(x)
|
|
|
608
|
|
|
609 typedef enum
|
|
|
610 {
|
|
|
611 SND_FORMAT_FORMAT_UNSPECIFIED,
|
|
|
612 SND_FORMAT_MULAW_8,
|
|
|
613 SND_FORMAT_LINEAR_8,
|
|
|
614 SND_FORMAT_LINEAR_16,
|
|
|
615 SND_FORMAT_LINEAR_24,
|
|
|
616 SND_FORMAT_LINEAR_32,
|
|
|
617 SND_FORMAT_FLOAT,
|
|
|
618 SND_FORMAT_DOUBLE,
|
|
|
619 SND_FORMAT_INDIRECT,
|
|
|
620 SND_FORMAT_NESTED,
|
|
|
621 SND_FORMAT_DSP_CODE,
|
|
|
622 SND_FORMAT_DSP_DATA_8,
|
|
|
623 SND_FORMAT_DSP_DATA_16,
|
|
|
624 SND_FORMAT_DSP_DATA_24,
|
|
|
625 SND_FORMAT_DSP_DATA_32,
|
|
|
626 SND_FORMAT_DSP_unknown_15,
|
|
|
627 SND_FORMAT_DISPLAY,
|
|
|
628 SND_FORMAT_MULAW_SQUELCH,
|
|
|
629 SND_FORMAT_EMPHASIZED,
|
|
|
630 SND_FORMAT_COMPRESSED,
|
|
|
631 SND_FORMAT_COMPRESSED_EMPHASIZED,
|
|
|
632 SND_FORMAT_DSP_COMMANDS,
|
|
|
633 SND_FORMAT_DSP_COMMANDS_SAMPLES
|
|
|
634 }
|
|
|
635 SNDFormatCode;
|
|
|
636
|
|
|
637 static int
|
|
|
638 parse_snd_header (void *header, long length, AudioContext desc)
|
|
|
639 {
|
|
|
640 #define hp ((SNDSoundStruct *) (header))
|
|
|
641 long limit;
|
|
|
642
|
|
|
643 #if HAVE_LINEAR
|
|
|
644 desc->ac_write_chunk_function = write_linear_chunk;
|
|
|
645 #endif
|
|
|
646 switch ((SNDFormatCode) SOUND_TO_HOST_INT (hp->dataFormat))
|
|
|
647 {
|
|
|
648 #if HAVE_MULAW_8
|
|
|
649 case SND_FORMAT_MULAW_8:
|
|
|
650 desc->ac_format = AFmulaw8;
|
|
|
651 desc->ac_write_chunk_function = write_mulaw_8_chunk;
|
|
|
652 break;
|
|
|
653 #endif
|
|
|
654 #if HAVE_LINEAR
|
|
|
655 case SND_FORMAT_LINEAR_8:
|
|
|
656 desc->ac_format = AFlinear8;
|
|
|
657 break;
|
|
|
658 case SND_FORMAT_LINEAR_16:
|
|
|
659 desc->ac_format = AFlinear16;
|
|
|
660 break;
|
|
|
661 case SND_FORMAT_LINEAR_24:
|
|
|
662 desc->ac_format = AFlinear24;
|
|
|
663 break;
|
|
|
664 #endif
|
|
|
665 #if HAVE_LINEAR_32
|
|
|
666 case SND_FORMAT_LINEAR_32:
|
|
|
667 desc->ac_format = AFlinear32;
|
|
|
668 desc->ac_write_chunk_function = write_linear_32_chunk;
|
|
|
669 break;
|
|
|
670 #endif
|
|
|
671 default:
|
|
|
672 desc->ac_format = AFunknown;
|
|
|
673 }
|
|
|
674 desc->ac_output_rate = SOUND_TO_HOST_INT (hp->samplingRate);
|
|
|
675 desc->ac_nchan = SOUND_TO_HOST_INT (hp->channelCount);
|
|
563
|
676 desc->ac_data = (Char_Binary *) header + SOUND_TO_HOST_INT (hp->dataLocation);
|
|
|
677 limit = (Char_Binary *) header + length - (Char_Binary *) desc->ac_data;
|
|
428
|
678 desc->ac_size = SOUND_TO_HOST_INT (hp->dataSize);
|
|
|
679 if (desc->ac_size > limit) desc->ac_size = limit;
|
|
|
680 return 0;
|
|
|
681 #undef hp
|
|
|
682 }
|
|
|
683 #endif /* HAVE_SND_FILES */
|
|
|
684
|
