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0
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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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24 #include <config.h>
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25 #include "lisp.h"
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26
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27 #include <audio.h>
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28 #include <sys/file.h>
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29 #include <sys/types.h>
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30 #include <sys/stat.h>
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31 #include <fcntl.h>
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367
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32 #include <unistd.h>
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0
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33 #include <string.h>
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34 #include <netinet/in.h> /* for ntohl() etc. */
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35
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36 /* Configuration options */
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37
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38 /* ability to parse Sun/NeXT (.au or .snd) audio file headers. The
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39 .snd format supports all sampling rates and sample widths that are
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40 commonly used, as well as stereo. It is also easy to parse. */
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41 #ifndef HAVE_SND_FILES
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42 #define HAVE_SND_FILES 1
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43 #endif
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44
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45 /* support for eight-but mu-law encoding. This is a useful compaction
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46 technique, and most sounds from the Sun universe are in this
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47 format. */
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48 #ifndef HAVE_MULAW_8
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49 #define HAVE_MULAW_8 1
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50 #endif
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51
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52 /* if your machine is very slow, you have to use a table lookup to
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53 convert mulaw samples to linear. This makes Emacs bigger so try to
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54 avoid it. */
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55 #ifndef USE_MULAW_DECODE_TABLE
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56 #define USE_MULAW_DECODE_TABLE 0
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57 #endif
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58
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59 /* support for linear encoding -- useful if you want better quality.
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60 This enables 8, 16 and 24 bit wide samples. */
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61 #ifndef HAVE_LINEAR
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62 #define HAVE_LINEAR 1
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63 #endif
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64
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65 /* support for 32 bit wide samples. If you notice the difference
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66 between 32 and 24 bit samples, you must have very good ears. Since
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67 the SGI audio library only supports 24 bit samples, each sample has
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68 to be shifted right by 8 bits anyway. So you should probably just
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69 convert all your 32 bit audio files to 24 bit. */
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70 #ifndef HAVE_LINEAR_32
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71 #define HAVE_LINEAR_32 0
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72 #endif
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73
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74 /* support for stereo sound. Imagine the cool applications of this:
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75 finally you don't just hear a beep -- you also know immediately
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76 *where* something went wrong! Unfortunately the programming
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77 interface only takes a single volume argument so far. */
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78 #ifndef HAVE_STEREO
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79 #define HAVE_STEREO 1
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80 #endif
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81
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82 /* the play routine can be interrupted between chunks, so we choose a
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83 small chunksize to keep the system responsive (2000 samples
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84 correspond to a quarter of a second for .au files. If you
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85 HAVE_STEREO, the chunksize should probably be even. */
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86 #define CHUNKSIZE 8000
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87
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88 /* the format assumed for header-less audio data. The following
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89 assumes ".au" format (8000 samples/sec mono 8-bit mulaw). */
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90 #define DEFAULT_SAMPLING_RATE 8000
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91 #define DEFAULT_CHANNEL_COUNT 1
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92 #define DEFAULT_FORMAT AFmulaw8
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93
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94 /* Exports */
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95
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96 /* all compilers on machines that have the SGI audio library
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97 understand prototypes, right? */
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98
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99 extern void play_sound_file (char *, int);
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100 extern void play_sound_data (unsigned char *, int, int);
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101
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102 /* Data structures */
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103
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104 /* an AudioContext describes everything we want to know about how a
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105 particular sound snippet should be played. It is split into three
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106 parts (device, port and buffer) for implementation reasons. The
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107 device part corresponds to the state of the output device and must
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108 be reverted after playing the samples. The port part corresponds
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109 to an ALport; we want to allocate a minimal number of these since
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110 there are only four of them system-wide, but on the other hand we
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111 can't use the same port for mono and stereo. The buffer part
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112 corresponds to the sound data itself. */
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113
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114 typedef struct _AudioContextRec * AudioContext;
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115
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116 typedef struct
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117 {
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118 long device;
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119 int left_speaker_gain;
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120 int right_speaker_gain;
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121 long output_rate;
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122 }
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123 AudioDeviceRec, * AudioDevice;
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124
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125 /* supported sound data formats */
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126
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127 typedef enum
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128 {
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129 AFunknown,
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130 #if HAVE_MULAW_8
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131 AFmulaw8,
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132 #endif
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133 #if HAVE_LINEAR
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134 AFlinear8,
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135 AFlinear16,
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136 AFlinear24,
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137 #if HAVE_LINEAR_32
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138 AFlinear32,
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139 #endif
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140 #endif
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141 AFillegal
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142 }
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143 AudioFormat;
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144
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145 typedef struct
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146 {
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147 ALport port;
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148 AudioFormat format;
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149 unsigned nchan;
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150 unsigned queue_size;
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151 }
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152 AudioPortRec, * AudioPort;
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153
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154 typedef struct
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155 {
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156 void * data;
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157 unsigned long size;
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158 void (* write_chunk_function) (void *, void *, AudioContext);
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159 }
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160 AudioBufferRec, * AudioBuffer;
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161
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162 typedef struct _AudioContextRec
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163 {
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164 AudioDeviceRec device;
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165 AudioPortRec port;
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166 AudioBufferRec buffer;
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167 }
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168 AudioContextRec;
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169
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170 #define ac_device device.device
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171 #define ac_left_speaker_gain device.left_speaker_gain
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172 #define ac_right_speaker_gain device.right_speaker_gain
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173 #define ac_output_rate device.output_rate
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174 #define ac_port port.port
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175 #define ac_format port.format
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176 #define ac_nchan port.nchan
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177 #define ac_queue_size port.queue_size
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178 #define ac_data buffer.data
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179 #define ac_size buffer.size
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180 #define ac_write_chunk_function buffer.write_chunk_function
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181
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182 /* Forward declarations */
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183
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184 static Lisp_Object close_sound_file (Lisp_Object);
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185 static AudioContext audio_initialize (unsigned char *, int, int);
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186 static void play_internal (unsigned char *, int, AudioContext);
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187 static void drain_audio_port (AudioContext);
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188 static void write_mulaw_8_chunk (void *, void *, AudioContext);
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189 static void write_linear_chunk (void *, void *, AudioContext);
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190 static void write_linear_32_chunk (void *, void *, AudioContext);
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191 static Lisp_Object restore_audio_port (Lisp_Object);
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192 static AudioContext initialize_audio_port (AudioContext);
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193 static int open_audio_port (AudioContext, AudioContext);
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194 static void adjust_audio_volume (AudioDevice);
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195 static void get_current_volumes (AudioDevice);
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196 static int set_channels (ALconfig, unsigned);
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197 static int set_output_format (ALconfig, AudioFormat);
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198 static int parse_snd_header (void*, long, AudioContext);
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199
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200 /* are we looking at an NeXT/Sun audio header? */
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201 #define LOOKING_AT_SND_HEADER_P(address) \
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202 (!strncmp(".snd", (char *)(address), 4))
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203
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204 static Lisp_Object
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205 close_sound_file (closure)
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206 Lisp_Object closure;
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207 {
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208 close (XINT (closure));
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209 return Qnil;
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210 }
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211
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212 void
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213 play_sound_file (sound_file, volume)
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214 char * sound_file;
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215 int volume;
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216 {
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217 int count = specpdl_depth ();
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218 int input_fd;
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219 unsigned char buffer[CHUNKSIZE];
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220 int bytes_read;
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221 AudioContext ac = (AudioContext) 0;
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222
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223 input_fd = open (sound_file, O_RDONLY);
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224 if (input_fd == -1)
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225 /* no error message -- this can't happen
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226 because Fplay_sound_file has checked the
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227 file for us. */
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228 return;
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229
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230 record_unwind_protect (close_sound_file, make_int (input_fd));
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231
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232 while ((bytes_read = read (input_fd, buffer, CHUNKSIZE)) > 0)
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233 {
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234 if (ac == (AudioContext) 0)
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235 {
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236 ac = audio_initialize (buffer, bytes_read, volume);
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237 if (ac == 0)
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238 return;
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239 }
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240 else
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241 {
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242 ac->ac_data = buffer;
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243 ac->ac_size = bytes_read;
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244 }
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245 play_internal (buffer, bytes_read, ac);
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246 }
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247 drain_audio_port (ac);
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248 unbind_to (count, Qnil);
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249 }
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250
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251 static long
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252 saved_device_state[] = {
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253 AL_OUTPUT_RATE, 0,
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254 AL_LEFT_SPEAKER_GAIN, 0,
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255 AL_RIGHT_SPEAKER_GAIN, 0,
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256 };
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257
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258 static Lisp_Object
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259 restore_audio_port (closure)
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260 Lisp_Object closure;
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261 {
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173
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262 Lisp_Object * contents = XVECTOR_DATA (closure);
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0
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263 saved_device_state[1] = XINT (contents[0]);
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264 saved_device_state[3] = XINT (contents[1]);
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265 saved_device_state[5] = XINT (contents[2]);
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266 ALsetparams (AL_DEFAULT_DEVICE, saved_device_state, 6);
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267 return Qnil;
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268 }
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269
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270 void
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271 play_sound_data (data, length, volume)
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272 unsigned char * data;
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273 int length;
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274 int volume;
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275 {
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276 int count = specpdl_depth ();
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277 AudioContext ac;
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278
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279 ac = audio_initialize (data, length, volume);
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280 if (ac == (AudioContext) 0)
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281 return;
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282 play_internal (data, length, ac);
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283 drain_audio_port (ac);
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284 unbind_to (count, Qnil);
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285 }
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286
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287 static AudioContext
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288 audio_initialize (data, length, volume)
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289 unsigned char * data;
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290 int length;
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291 int volume;
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292 {
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293 Lisp_Object audio_port_state[3];
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294 static AudioContextRec desc;
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295 AudioContext ac;
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296
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297 desc.ac_right_speaker_gain
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298 = desc.ac_left_speaker_gain
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299 = volume * 256 / 100;
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300 desc.ac_device = AL_DEFAULT_DEVICE;
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301
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302 #if HAVE_SND_FILES
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303 if (LOOKING_AT_SND_HEADER_P (data))
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304 {
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305 if (parse_snd_header (data, length, & desc)==-1)
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306 report_file_error ("decoding .snd header", Qnil);
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307 }
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308 else
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309 #endif
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310 {
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311 desc.ac_data = data;
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312 desc.ac_size = length;
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313 desc.ac_output_rate = DEFAULT_SAMPLING_RATE;
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314 desc.ac_nchan = DEFAULT_CHANNEL_COUNT;
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315 desc.ac_format = DEFAULT_FORMAT;
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316 desc.ac_write_chunk_function = write_mulaw_8_chunk;
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317 }
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318
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319 /* Make sure that the audio port is reset to
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320 its initial characteristics after exit */
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321 ALgetparams (desc.ac_device, saved_device_state,
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322 sizeof (saved_device_state) / sizeof (long));
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323 audio_port_state[0] = make_int (saved_device_state[1]);
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324 audio_port_state[1] = make_int (saved_device_state[3]);
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325 audio_port_state[2] = make_int (saved_device_state[5]);
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326 record_unwind_protect (restore_audio_port,
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327 Fvector (3, &audio_port_state[0]));
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185
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328
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0
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329 ac = initialize_audio_port (& desc);
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330 desc = * ac;
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331 return ac;
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332 }
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333
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334 static void
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335 play_internal (data, length, ac)
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336 unsigned char * data;
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337 int length;
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338 AudioContext ac;
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339 {
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340 unsigned char * limit;
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341 if (ac == (AudioContext) 0)
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342 return;
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343
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344 data = ac->ac_data;
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345 limit = data + ac->ac_size;
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346 while (data < limit)
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347 {
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348 unsigned char * chunklimit = data + CHUNKSIZE;
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349
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350 if (chunklimit > limit)
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351 chunklimit = limit;
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352
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353 QUIT;
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354
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355 (* ac->ac_write_chunk_function) (data, chunklimit, ac);
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356 data = chunklimit;
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357 }
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358 }
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359
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360 static void
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361 drain_audio_port (ac)
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362 AudioContext ac;
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363 {
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364 while (ALgetfilled (ac->ac_port) > 0)
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365 sginap(1);
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366 }
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367
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368 /* Methods to write a "chunk" from a buffer containing audio data to
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369 an audio port. This may involve some conversion if the output
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370 device doesn't directly support the format the audio data is in. */
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371
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372 #if HAVE_MULAW_8
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373
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374 #if USE_MULAW_DECODE_TABLE
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375 #include "libst.h"
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376 #else /* not USE_MULAW_DECODE_TABLE */
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377 static int
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378 st_ulaw_to_linear (u)
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379 int u;
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380 {
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381 static CONST short table[] = {0,132,396,924,1980,4092,8316,16764};
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382 int u1 = ~u;
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383 short exponent = (u1 >> 4) & 0x07;
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384 int mantissa = u1 & 0x0f;
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385 int unsigned_result = table[exponent]+(mantissa << (exponent+3));
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386 return u1 & 0x80 ? -unsigned_result : unsigned_result;
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387 }
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388 #endif /* not USE_MULAW_DECODE_TABLE */
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389
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390 static void
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391 write_mulaw_8_chunk (buffer, chunklimit, ac)
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392 void * buffer;
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393 void * chunklimit;
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394 AudioContext ac;
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395 {
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396 unsigned char * data = (unsigned char *) buffer;
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397 unsigned char * limit = (unsigned char *) chunklimit;
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398 short * obuf, * bufp;
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399 long n_samples = limit - data;
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400
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185
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401 obuf = alloca_array (short, n_samples);
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0
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402 bufp = &obuf[0];
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403
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404 while (data < limit)
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405 *bufp++ = st_ulaw_to_linear (*data++);
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406 ALwritesamps (ac->ac_port, obuf, n_samples);
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407 }
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408 #endif /* HAVE_MULAW_8 */
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409
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410 #if HAVE_LINEAR
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411 static void
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412 write_linear_chunk (data, limit, ac)
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413 void * data;
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414 void * limit;
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415 AudioContext ac;
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416 {
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417 unsigned n_samples;
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418
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419 switch (ac->ac_format)
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420 {
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421 case AFlinear16: n_samples = (short *) limit - (short *) data; break;
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422 case AFlinear8: n_samples = (char *) limit - (char *) data; break;
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423 default: n_samples = (long *) limit - (long *) data; break;
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424 }
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|
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425 ALwritesamps (ac->ac_port, data, (long) n_samples);
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|
|
426 }
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|
|
427
|
|
|
428 #if HAVE_LINEAR_32
|
|
|
429 static void
|
|
|
430 write_linear_32_chunk (buffer, chunklimit, ac)
|
|
|
431 void * buffer;
|
|
|
432 void * chunklimit;
|
|
|
433 AudioContext ac;
|
|
|
434 {
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|
|
435 long * data = (long *) buffer;
|
|
|
436 long * limit = (long *) chunklimit;
|
|
|
437 long * obuf, * bufp;
|
|
|
438 long n_samples = limit-data;
|
|
|
439
|
|
185
|
440 obuf = alloca_array (long, n_samples);
|
|
0
|
441 bufp = &obuf[0];
|
|
|
442
|
|
|
443 while (data < limit)
|
|
|
444 *bufp++ = *data++ >> 8;
|
|
|
445 ALwritesamps (ac->ac_port, obuf, n_samples);
|
|
|
446 }
|
|
|
447 #endif /* HAVE_LINEAR_32 */
|
|
|
448 #endif /* HAVE_LINEAR */
|
|
|
449
|
|
|
450 static AudioContext
|
|
|
451 initialize_audio_port (desc)
|
|
|
452 AudioContext desc;
|
|
|
453 {
|
|
|
454 /* we can't use the same port for mono and stereo */
|
|
|
455 static AudioContextRec mono_port_state
|
|
|
456 = { { 0, 0, 0, 0 },
|
|
|
457 { (ALport) 0, AFunknown, 1, 0 },
|
|
|
458 { (void *) 0, (unsigned long) 0 } };
|
|
|
459 #if HAVE_STEREO
|
|
|
460 static AudioContextRec stereo_port_state
|
|
|
461 = { { 0, 0, 0, 0 },
|
|
|
462 { (ALport) 0, AFunknown, 2, 0 },
|
|
|
463 { (void *) 0, (unsigned long) 0 } };
|
|
|
464 static AudioContext return_ac;
|
|
|
465
|
|
|
466 switch (desc->ac_nchan)
|
|
|
467 {
|
|
|
468 case 1: return_ac = & mono_port_state; break;
|
|
|
469 case 2: return_ac = & stereo_port_state; break;
|
|
|
470 default: return (AudioContext) 0;
|
|
|
471 }
|
|
|
472 #else /* not HAVE_STEREO */
|
|
|
473 static AudioContext return_ac = & mono_port_state;
|
|
|
474 #endif /* not HAVE_STEREO */
|
|
|
475
|
|
|
476 return_ac->device = desc->device;
|
|
|
477 return_ac->buffer = desc->buffer;
|
|
|
478 return_ac->ac_format = desc->ac_format;
|
|
|
479 return_ac->ac_queue_size = desc->ac_queue_size;
|
|
|
480
|
|
|
481 if (return_ac->ac_port==(ALport) 0)
|
|
|
482 {
|
|
|
483 if ((open_audio_port (return_ac, desc))==-1)
|
|
|
484 {
|
|
|
485 report_file_error ("Open audio port", Qnil);
|
|
|
486 return (AudioContext) 0;
|
|
|
487 }
|
|
|
488 }
|
|
|
489 else
|
|
|
490 {
|
|
|
491 ALconfig config = ALgetconfig (return_ac->ac_port);
|
|
|
492 int changed = 0;
|
|
|
493 long params[2];
|
|
|
494
|
|
|
495 params[0] = AL_OUTPUT_RATE;
|
|
|
496 ALgetparams (return_ac->ac_device, params, 2);
|
|
|
497 return_ac->ac_output_rate = params[1];
|
|
|
498
|
|
|
499 if (return_ac->ac_output_rate != desc->ac_output_rate)
|
|
|
500 {
|
|
|
501 return_ac->ac_output_rate = params[1] = desc->ac_output_rate;
|
|
|
502 ALsetparams (return_ac->ac_device, params, 2);
|
|
|
503 }
|
|
|
504 if ((changed = set_output_format (config, return_ac->ac_format))==-1)
|
|
|
505 return (AudioContext) 0;
|
|
|
506 return_ac->ac_format = desc->ac_format;
|
|
|
507 if (changed)
|
|
|
508 ALsetconfig (return_ac->ac_port, config);
|
|
|
509 }
|
|
|
510 return_ac->ac_write_chunk_function = desc->ac_write_chunk_function;
|
|
|
511 get_current_volumes (& return_ac->device);
|
|
|
512 if (return_ac->ac_left_speaker_gain != desc->ac_left_speaker_gain
|
|
|
513 || return_ac->ac_right_speaker_gain != desc->ac_right_speaker_gain)
|
|
|
514 adjust_audio_volume (& desc->device);
|
|
|
515 return return_ac;
|
|
|
516 }
|
|
|
517
|
|
|
518 static int
|
|
|
519 open_audio_port (return_ac, desc)
|
|
|
520 AudioContext return_ac;
|
|
|
521 AudioContext desc;
|
|
|
522 {
|
|
|
523 ALconfig config = ALnewconfig();
|
|
|
524 long params[2];
|
|
|
525
|
|
|
526 adjust_audio_volume (& desc->device);
|
|
|
527 return_ac->ac_left_speaker_gain = desc->ac_left_speaker_gain;
|
|
|
528 return_ac->ac_right_speaker_gain = desc->ac_right_speaker_gain;
|
|
|
529 params[0] = AL_OUTPUT_RATE;
|
|
|
530 params[1] = desc->ac_output_rate;
|
|
|
531 ALsetparams (desc->ac_device, params, 2);
|
|
|
532 return_ac->ac_output_rate = desc->ac_output_rate;
|
|
|
533 if (set_channels (config, desc->ac_nchan)==-1)
|
|
|
534 return -1;
|
|
|
535 return_ac->ac_nchan = desc->ac_nchan;
|
|
|
536 if (set_output_format (config, desc->ac_format)==-1)
|
|
|
537 return -1;
|
|
|
538 return_ac->ac_format = desc->ac_format;
|
|
|
539 ALsetqueuesize (config, (long) CHUNKSIZE);
|
|
|
540 return_ac->ac_port = ALopenport("XEmacs audio output", "w", config);
|
|
|
541 ALfreeconfig (config);
|
|
|
542 if (return_ac->ac_port==0)
|
|
|
543 {
|
|
|
544 report_file_error ("Opening audio output port", Qnil);
|
|
|
545 return -1;
|
|
|
546 }
|
|
|
547 return 0;
|
|
|
548 }
|
|
|
549
|
|
|
550 static int
|
|
|
551 set_channels (config, nchan)
|
|
|
552 ALconfig config;
|
|
|
553 unsigned nchan;
|
|
|
554 {
|
|
|
555 switch (nchan)
|
|
|
556 {
|
|
|
557 case 1: ALsetchannels (config, AL_MONO); break;
|
|
|
558 #if HAVE_STEREO
|
|
|
559 case 2: ALsetchannels (config, AL_STEREO); break;
|
|
|
560 #endif /* HAVE_STEREO */
|
|
|
561 default:
|
|
|
562 report_file_error ("Unsupported channel count",
|
|
|
563 Fcons (make_int (nchan), Qnil));
|
|
|
564 return -1;
|
|
|
565 }
|
|
|
566 return 0;
|
|
|
567 }
|
|
|
568
|
|
|
569 static int
|
|
|
570 set_output_format (config, format)
|
|
|
571 ALconfig config;
|
|
|
572 AudioFormat format;
|
|
|
573 {
|
|
|
574 long samplesize;
|
|
|
575 long old_samplesize;
|
|
|
576
|
|
|
577 switch (format)
|
|
|
578 {
|
|
|
579 #if HAVE_MULAW_8
|
|
|
580 case AFmulaw8:
|
|
|
581 #endif
|
|
|
582 #if HAVE_LINEAR
|
|
|
583 case AFlinear16:
|
|
|
584 #endif
|
|
|
585 #if HAVE_MULAW_8 || HAVE_LINEAR
|
|
|
586 samplesize = AL_SAMPLE_16;
|
|
|
587 break;
|
|
|
588 #endif
|
|
|
589 #if HAVE_LINEAR
|
|
|
590 case AFlinear8:
|
|
|
591 samplesize = AL_SAMPLE_8;
|
|
|
592 break;
|
|
|
593 case AFlinear24:
|
|
|
594 #if HAVE_LINEAR_32
|
|
|
595 case AFlinear32:
|
|
|
596 samplesize = AL_SAMPLE_24;
|
|
|
597 break;
|
|
|
598 #endif
|
|
|
599 #endif
|
|
|
600 default:
|
|
|
601 report_file_error ("Unsupported audio format",
|
|
|
602 Fcons (make_int (format), Qnil));
|
|
|
603 return -1;
|
|
|
604 }
|
|
|
605 old_samplesize = ALgetwidth (config);
|
|
|
606 if (old_samplesize==samplesize)
|
|
|
607 return 0;
|
|
|
608 ALsetwidth (config, samplesize);
|
|
|
609 return 1;
|
|
|
610 }
|
|
|
611
|
|
|
612 static void
|
|
|
613 adjust_audio_volume (device)
|
|
|
614 AudioDevice device;
|
|
|
615 {
|
|
|
616 long params[4];
|
|
|
617 params[0] = AL_LEFT_SPEAKER_GAIN;
|
|
|
618 params[1] = device->left_speaker_gain;
|
|
|
619 params[2] = AL_RIGHT_SPEAKER_GAIN;
|
|
|
620 params[3] = device->right_speaker_gain;
|
|
|
621 ALsetparams (device->device, params, 4);
|
|
|
622 }
|
|
|
623
|
|
|
624 static void
|
|
|
625 get_current_volumes (device)
|
|
|
626 AudioDevice device;
|
|
|
627 {
|
|
|
628 long params[4];
|
|
|
629 params[0] = AL_LEFT_SPEAKER_GAIN;
|
|
|
630 params[2] = AL_RIGHT_SPEAKER_GAIN;
|
|
|
631 ALgetparams (device->device, params, 4);
|
|
|
632 device->left_speaker_gain = params[1];
|
|
|
633 device->right_speaker_gain = params[3];
|
|
|
634 }
|
|
|
635
|
|
|
636 #if HAVE_SND_FILES
|
|
|
637
|
|
|
638 /* Parsing .snd (NeXT/Sun) headers */
|
|
|
639
|
|
|
640 typedef struct
|
|
|
641 {
|
|
|
642 int magic;
|
|
|
643 int dataLocation;
|
|
|
644 int dataSize;
|
|
|
645 int dataFormat;
|
|
|
646 int samplingRate;
|
|
|
647 int channelCount;
|
|
|
648 char info[4];
|
|
|
649 }
|
|
|
650 SNDSoundStruct;
|
|
|
651 #define SOUND_TO_HOST_INT(x) ntohl(x)
|
|
|
652
|
|
|
653 typedef enum
|
|
|
654 {
|
|
|
655 SND_FORMAT_FORMAT_UNSPECIFIED,
|
|
|
656 SND_FORMAT_MULAW_8,
|
|
|
657 SND_FORMAT_LINEAR_8,
|
|
|
658 SND_FORMAT_LINEAR_16,
|
|
|
659 SND_FORMAT_LINEAR_24,
|
|
|
660 SND_FORMAT_LINEAR_32,
|
|
|
661 SND_FORMAT_FLOAT,
|
|
|
662 SND_FORMAT_DOUBLE,
|
|
|
663 SND_FORMAT_INDIRECT,
|
|
|
664 SND_FORMAT_NESTED,
|
|
|
665 SND_FORMAT_DSP_CODE,
|
|
|
666 SND_FORMAT_DSP_DATA_8,
|
|
|
667 SND_FORMAT_DSP_DATA_16,
|
|
|
668 SND_FORMAT_DSP_DATA_24,
|
|
|
669 SND_FORMAT_DSP_DATA_32,
|
|
|
670 SND_FORMAT_DSP_unknown_15,
|
|
|
671 SND_FORMAT_DISPLAY,
|
|
|
672 SND_FORMAT_MULAW_SQUELCH,
|
|
|
673 SND_FORMAT_EMPHASIZED,
|
|
|
674 SND_FORMAT_COMPRESSED,
|
|
|
675 SND_FORMAT_COMPRESSED_EMPHASIZED,
|
|
|
676 SND_FORMAT_DSP_COMMANDS,
|
|
|
677 SND_FORMAT_DSP_COMMANDS_SAMPLES
|
|
|
678 }
|
|
|
679 SNDFormatCode;
|
|
|
680
|
|
|
681 static int
|
|
|
682 parse_snd_header (header, length, desc)
|
|
|
683 void * header;
|
|
|
684 long length;
|
|
|
685 AudioContext desc;
|
|
|
686 {
|
|
|
687 #define hp ((SNDSoundStruct *) (header))
|
|
|
688 long limit;
|
|
|
689
|
|
|
690 #if HAVE_LINEAR
|
|
|
691 desc->ac_write_chunk_function = write_linear_chunk;
|
|
|
692 #endif
|
|
|
693 switch ((SNDFormatCode) SOUND_TO_HOST_INT (hp->dataFormat))
|
|
|
694 {
|
|
|
695 #if HAVE_MULAW_8
|
|
|
696 case SND_FORMAT_MULAW_8:
|
|
|
697 desc->ac_format = AFmulaw8;
|
|
|
698 desc->ac_write_chunk_function = write_mulaw_8_chunk;
|
|
|
699 break;
|
|
|
700 #endif
|
|
|
701 #if HAVE_LINEAR
|
|
|
702 case SND_FORMAT_LINEAR_8:
|
|
|
703 desc->ac_format = AFlinear8;
|
|
|
704 break;
|
|
|
705 case SND_FORMAT_LINEAR_16:
|
|
|
706 desc->ac_format = AFlinear16;
|
|
|
707 break;
|
|
|
708 case SND_FORMAT_LINEAR_24:
|
|
|
709 desc->ac_format = AFlinear24;
|
|
|
710 break;
|
|
|
711 #endif
|
|
|
712 #if HAVE_LINEAR_32
|
|
|
713 case SND_FORMAT_LINEAR_32:
|
|
|
714 desc->ac_format = AFlinear32;
|
|
|
715 desc->ac_write_chunk_function = write_linear_32_chunk;
|
|
|
716 break;
|
|
|
717 #endif
|
|
|
718 default:
|
|
|
719 desc->ac_format = AFunknown;
|
|
|
720 }
|
|
|
721 desc->ac_output_rate = SOUND_TO_HOST_INT (hp->samplingRate);
|
|
|
722 desc->ac_nchan = SOUND_TO_HOST_INT (hp->channelCount);
|
|
|
723 desc->ac_data = (char *) header + SOUND_TO_HOST_INT (hp->dataLocation);
|
|
|
724 limit = (char *) header + length - (char *) desc->ac_data;
|
|
|
725 desc->ac_size = SOUND_TO_HOST_INT (hp->dataSize);
|
|
|
726 if (desc->ac_size > limit) desc->ac_size = limit;
|
|
|
727 return 0;
|
|
|
728 #undef hp
|
|
|
729 }
|
|
|
730 #endif /* HAVE_SND_FILES */
|
|
|
731
|
