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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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609
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34 #include <audio.h>
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35
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428
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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 /* Data structures */
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95
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96 /* an AudioContext describes everything we want to know about how a
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97 particular sound snippet should be played. It is split into three
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98 parts (device, port and buffer) for implementation reasons. The
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99 device part corresponds to the state of the output device and must
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100 be reverted after playing the samples. The port part corresponds
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101 to an ALport; we want to allocate a minimal number of these since
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102 there are only four of them system-wide, but on the other hand we
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103 can't use the same port for mono and stereo. The buffer part
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104 corresponds to the sound data itself. */
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105
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106 typedef struct _AudioContextRec * AudioContext;
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107
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108 typedef struct
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109 {
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110 long device;
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111 int left_speaker_gain;
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112 int right_speaker_gain;
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113 long output_rate;
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114 }
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115 AudioDeviceRec, * AudioDevice;
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116
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117 /* supported sound data formats */
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118
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119 typedef enum
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120 {
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121 AFunknown,
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122 #if HAVE_MULAW_8
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123 AFmulaw8,
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124 #endif
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125 #if HAVE_LINEAR
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126 AFlinear8,
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127 AFlinear16,
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128 AFlinear24,
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129 #if HAVE_LINEAR_32
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130 AFlinear32,
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131 #endif
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132 #endif
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133 AFillegal
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134 }
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135 AudioFormat;
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136
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137 typedef struct
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138 {
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139 ALport port;
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140 AudioFormat format;
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141 unsigned nchan;
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142 unsigned queue_size;
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143 }
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144 AudioPortRec, * AudioPort;
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145
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146 typedef struct
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147 {
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148 void * data;
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149 unsigned long size;
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150 void (* write_chunk_function) (void *, void *, AudioContext);
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151 }
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152 AudioBufferRec, * AudioBuffer;
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153
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154 typedef struct _AudioContextRec
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155 {
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156 AudioDeviceRec device;
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157 AudioPortRec port;
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158 AudioBufferRec buffer;
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159 }
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160 AudioContextRec;
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161
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162 #define ac_device device.device
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163 #define ac_left_speaker_gain device.left_speaker_gain
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164 #define ac_right_speaker_gain device.right_speaker_gain
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165 #define ac_output_rate device.output_rate
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166 #define ac_port port.port
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167 #define ac_format port.format
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168 #define ac_nchan port.nchan
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169 #define ac_queue_size port.queue_size
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170 #define ac_data buffer.data
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171 #define ac_size buffer.size
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172 #define ac_write_chunk_function buffer.write_chunk_function
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173
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174 /* Forward declarations */
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175
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176 static Lisp_Object close_sound_file (Lisp_Object);
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563
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177 static AudioContext audio_initialize (UChar_Binary *, int, int);
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609
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178 static int play_internal (UChar_Binary *, int, AudioContext);
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428
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179 static void drain_audio_port (AudioContext);
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180 static void write_mulaw_8_chunk (void *, void *, AudioContext);
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181 static void write_linear_chunk (void *, void *, AudioContext);
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182 static void write_linear_32_chunk (void *, void *, AudioContext);
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183 static Lisp_Object restore_audio_port (Lisp_Object);
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184 static AudioContext initialize_audio_port (AudioContext);
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185 static int open_audio_port (AudioContext, AudioContext);
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186 static void adjust_audio_volume (AudioDevice);
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187 static void get_current_volumes (AudioDevice);
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188 static int set_channels (ALconfig, unsigned);
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189 static int set_output_format (ALconfig, AudioFormat);
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190 static int parse_snd_header (void*, long, AudioContext);
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191
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192 /* are we looking at an NeXT/Sun audio header? */
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193 #define LOOKING_AT_SND_HEADER_P(address) \
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194 (!strncmp(".snd", (char *)(address), 4))
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195
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196 static Lisp_Object
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197 close_sound_file (Lisp_Object closure)
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198 {
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199 close (XINT (closure));
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200 return Qnil;
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201 }
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202
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203 void
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563
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204 play_sound_file (Extbyte *sound_file, int volume)
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428
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205 {
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206 int count = specpdl_depth ();
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207 int input_fd;
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563
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208 UChar_Binary buffer[CHUNKSIZE];
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428
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209 int bytes_read;
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210 AudioContext ac = (AudioContext) 0;
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211
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212 input_fd = open (sound_file, O_RDONLY);
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213 if (input_fd == -1)
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214 /* no error message -- this can't happen
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215 because Fplay_sound_file has checked the
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216 file for us. */
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217 return;
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218
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219 record_unwind_protect (close_sound_file, make_int (input_fd));
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220
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221 while ((bytes_read = read (input_fd, buffer, CHUNKSIZE)) > 0)
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222 {
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223 if (ac == (AudioContext) 0)
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224 {
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225 ac = audio_initialize (buffer, bytes_read, volume);
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226 if (ac == 0)
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227 return;
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228 }
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229 else
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230 {
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231 ac->ac_data = buffer;
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232 ac->ac_size = bytes_read;
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233 }
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234 play_internal (buffer, bytes_read, ac);
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235 }
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236 drain_audio_port (ac);
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771
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237 unbind_to (count);
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428
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238 }
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239
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240 static long
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241 saved_device_state[] = {
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242 AL_OUTPUT_RATE, 0,
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243 AL_LEFT_SPEAKER_GAIN, 0,
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244 AL_RIGHT_SPEAKER_GAIN, 0,
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245 };
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246
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247 static Lisp_Object
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248 restore_audio_port (Lisp_Object closure)
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249 {
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250 Lisp_Object * contents = XVECTOR_DATA (closure);
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251 saved_device_state[1] = XINT (contents[0]);
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252 saved_device_state[3] = XINT (contents[1]);
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253 saved_device_state[5] = XINT (contents[2]);
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254 ALsetparams (AL_DEFAULT_DEVICE, saved_device_state, 6);
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255 return Qnil;
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256 }
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257
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609
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258 int
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563
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259 play_sound_data (UChar_Binary *data, int length, int volume)
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428
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260 {
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261 int count = specpdl_depth ();
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262 AudioContext ac;
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609
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263 int result;
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428
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264
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265 ac = audio_initialize (data, length, volume);
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266 if (ac == (AudioContext) 0)
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609
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267 return 0;
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268 result = play_internal (data, length, ac);
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428
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269 drain_audio_port (ac);
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771
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270 unbind_to (count);
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609
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271 return result;
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428
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272 }
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273
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274 static AudioContext
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563
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275 audio_initialize (UChar_Binary *data, int length, int volume)
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428
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276 {
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277 Lisp_Object audio_port_state[3];
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278 static AudioContextRec desc;
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279 AudioContext ac;
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280
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281 desc.ac_right_speaker_gain
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282 = desc.ac_left_speaker_gain
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283 = volume * 256 / 100;
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284 desc.ac_device = AL_DEFAULT_DEVICE;
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285
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286 #if HAVE_SND_FILES
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287 if (LOOKING_AT_SND_HEADER_P (data))
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288 {
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289 if (parse_snd_header (data, length, & desc)==-1)
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563
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290 report_sound_error ("decoding .snd header", Qunbound);
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428
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291 }
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292 else
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293 #endif
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294 {
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295 desc.ac_data = data;
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296 desc.ac_size = length;
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297 desc.ac_output_rate = DEFAULT_SAMPLING_RATE;
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298 desc.ac_nchan = DEFAULT_CHANNEL_COUNT;
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299 desc.ac_format = DEFAULT_FORMAT;
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300 desc.ac_write_chunk_function = write_mulaw_8_chunk;
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301 }
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302
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303 /* Make sure that the audio port is reset to
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304 its initial characteristics after exit */
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305 ALgetparams (desc.ac_device, saved_device_state,
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306 sizeof (saved_device_state) / sizeof (long));
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307 audio_port_state[0] = make_int (saved_device_state[1]);
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308 audio_port_state[1] = make_int (saved_device_state[3]);
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309 audio_port_state[2] = make_int (saved_device_state[5]);
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310 record_unwind_protect (restore_audio_port,
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311 Fvector (3, &audio_port_state[0]));
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312
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313 ac = initialize_audio_port (& desc);
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314 desc = * ac;
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315 return ac;
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316 }
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317
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609
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318 static int
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563
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319 play_internal (UChar_Binary *data, int length, AudioContext ac)
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428
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320 {
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563
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321 UChar_Binary * limit;
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428
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322 if (ac == (AudioContext) 0)
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609
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323 return 0;
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428
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324
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563
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325 data = (UChar_Binary *) ac->ac_data;
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428
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326 limit = data + ac->ac_size;
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327 while (data < limit)
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328 {
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563
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329 UChar_Binary * chunklimit = data + CHUNKSIZE;
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428
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330
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331 if (chunklimit > limit)
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332 chunklimit = limit;
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333
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334 QUIT;
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335
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336 (* ac->ac_write_chunk_function) (data, chunklimit, ac);
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337 data = chunklimit;
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338 }
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609
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339
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340 return 1;
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428
|
341 }
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342
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|
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343 static void
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344 drain_audio_port (AudioContext ac)
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345 {
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346 while (ALgetfilled (ac->ac_port) > 0)
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347 sginap(1);
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|
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348 }
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349
|
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350 /* Methods to write a "chunk" from a buffer containing audio data to
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351 an audio port. This may involve some conversion if the output
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352 device doesn't directly support the format the audio data is in. */
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353
|
|
|
354 #if HAVE_MULAW_8
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|
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355
|
|
|
356 #if USE_MULAW_DECODE_TABLE
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|
357 #include "libst.h"
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|
|
358 #else /* not USE_MULAW_DECODE_TABLE */
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|
|
359 static int
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|
|
360 st_ulaw_to_linear (int u)
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|
|
361 {
|
|
442
|
362 static const short table[] = {0,132,396,924,1980,4092,8316,16764};
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|
428
|
363 int u1 = ~u;
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|
|
364 short exponent = (u1 >> 4) & 0x07;
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|
|
365 int mantissa = u1 & 0x0f;
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|
|
366 int unsigned_result = table[exponent]+(mantissa << (exponent+3));
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|
|
367 return u1 & 0x80 ? -unsigned_result : unsigned_result;
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|
|
368 }
|
|
|
369 #endif /* not USE_MULAW_DECODE_TABLE */
|
|
|
370
|
|
|
371 static void
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|
|
372 write_mulaw_8_chunk (void *buffer, void *chunklimit, AudioContext ac)
|
|
|
373 {
|
|
563
|
374 UChar_Binary * data = (UChar_Binary *) buffer;
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|
|
375 UChar_Binary * limit = (UChar_Binary *) chunklimit;
|
|
428
|
376 short * obuf, * bufp;
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|
|
377 long n_samples = limit - data;
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|
|
378
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|
|
379 obuf = alloca_array (short, n_samples);
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|
|
380 bufp = &obuf[0];
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|
381
|
|
|
382 while (data < limit)
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|
|
383 *bufp++ = st_ulaw_to_linear (*data++);
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|
|
384 ALwritesamps (ac->ac_port, obuf, n_samples);
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|
|
385 }
|
|
|
386 #endif /* HAVE_MULAW_8 */
|
|
|
387
|
|
|
388 #if HAVE_LINEAR
|
|
|
389 static void
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|
|
390 write_linear_chunk (void *data, void *limit, AudioContext ac)
|
|
|
391 {
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|
|
392 unsigned n_samples;
|
|
|
393
|
|
|
394 switch (ac->ac_format)
|
|
|
395 {
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|
|
396 case AFlinear16: n_samples = (short *) limit - (short *) data; break;
|
|
563
|
397 case AFlinear8: n_samples = (Char_Binary *) limit - (Char_Binary *) data; break;
|
|
428
|
398 default: n_samples = (long *) limit - (long *) data; break;
|
|
|
399 }
|
|
|
400 ALwritesamps (ac->ac_port, data, (long) n_samples);
|
|
|
401 }
|
|
|
402
|
|
|
403 #if HAVE_LINEAR_32
|
|
|
404 static void
|
|
|
405 write_linear_32_chunk (void *buffer, void *chunklimit, AudioContext ac)
|
|
|
406 {
|
|
|
407 long * data = (long *) buffer;
|
|
|
408 long * limit = (long *) chunklimit;
|
|
|
409 long * obuf, * bufp;
|
|
|
410 long n_samples = limit-data;
|
|
|
411
|
|
|
412 obuf = alloca_array (long, n_samples);
|
|
|
413 bufp = &obuf[0];
|
|
|
414
|
|
|
415 while (data < limit)
|
|
|
416 *bufp++ = *data++ >> 8;
|
|
|
417 ALwritesamps (ac->ac_port, obuf, n_samples);
|
|
|
418 }
|
|
|
419 #endif /* HAVE_LINEAR_32 */
|
|
|
420 #endif /* HAVE_LINEAR */
|
|
|
421
|
|
|
422 static AudioContext
|
|
|
423 initialize_audio_port (AudioContext desc)
|
|
|
424 {
|
|
|
425 /* we can't use the same port for mono and stereo */
|
|
|
426 static AudioContextRec mono_port_state
|
|
|
427 = { { 0, 0, 0, 0 },
|
|
|
428 { (ALport) 0, AFunknown, 1, 0 },
|
|
|
429 { (void *) 0, (unsigned long) 0 } };
|
|
|
430 #if HAVE_STEREO
|
|
|
431 static AudioContextRec stereo_port_state
|
|
|
432 = { { 0, 0, 0, 0 },
|
|
|
433 { (ALport) 0, AFunknown, 2, 0 },
|
|
|
434 { (void *) 0, (unsigned long) 0 } };
|
|
|
435 static AudioContext return_ac;
|
|
|
436
|
|
|
437 switch (desc->ac_nchan)
|
|
|
438 {
|
|
|
439 case 1: return_ac = & mono_port_state; break;
|
|
|
440 case 2: return_ac = & stereo_port_state; break;
|
|
|
441 default: return (AudioContext) 0;
|
|
|
442 }
|
|
|
443 #else /* not HAVE_STEREO */
|
|
|
444 static AudioContext return_ac = & mono_port_state;
|
|
|
445 #endif /* not HAVE_STEREO */
|
|
|
446
|
|
|
447 return_ac->device = desc->device;
|
|
|
448 return_ac->buffer = desc->buffer;
|
|
|
449 return_ac->ac_format = desc->ac_format;
|
|
|
450 return_ac->ac_queue_size = desc->ac_queue_size;
|
|
|
451
|
|
|
452 if (return_ac->ac_port==(ALport) 0)
|
|
|
453 {
|
|
|
454 if ((open_audio_port (return_ac, desc))==-1)
|
|
|
455 {
|
|
563
|
456 report_sound_error ("Open audio port", Qunbound);
|
|
428
|
457 return (AudioContext) 0;
|
|
|
458 }
|
|
|
459 }
|
|
|
460 else
|
|
|
461 {
|
|
|
462 ALconfig config = ALgetconfig (return_ac->ac_port);
|
|
|
463 int changed = 0;
|
|
|
464 long params[2];
|
|
|
465
|
|
|
466 params[0] = AL_OUTPUT_RATE;
|
|
|
467 ALgetparams (return_ac->ac_device, params, 2);
|
|
|
468 return_ac->ac_output_rate = params[1];
|
|
|
469
|
|
|
470 if (return_ac->ac_output_rate != desc->ac_output_rate)
|
|
|
471 {
|
|
|
472 return_ac->ac_output_rate = params[1] = desc->ac_output_rate;
|
|
|
473 ALsetparams (return_ac->ac_device, params, 2);
|
|
|
474 }
|
|
|
475 if ((changed = set_output_format (config, return_ac->ac_format))==-1)
|
|
|
476 return (AudioContext) 0;
|
|
|
477 return_ac->ac_format = desc->ac_format;
|
|
|
478 if (changed)
|
|
|
479 ALsetconfig (return_ac->ac_port, config);
|
|
|
480 }
|
|
|
481 return_ac->ac_write_chunk_function = desc->ac_write_chunk_function;
|
|
|
482 get_current_volumes (& return_ac->device);
|
|
|
483 if (return_ac->ac_left_speaker_gain != desc->ac_left_speaker_gain
|
|
|
484 || return_ac->ac_right_speaker_gain != desc->ac_right_speaker_gain)
|
|
|
485 adjust_audio_volume (& desc->device);
|
|
|
486 return return_ac;
|
|
|
487 }
|
|
|
488
|
|
|
489 static int
|
|
|
490 open_audio_port (AudioContext return_ac, AudioContext desc)
|
|
|
491 {
|
|
|
492 ALconfig config = ALnewconfig();
|
|
|
493 long params[2];
|
|
|
494
|
|
|
495 adjust_audio_volume (& desc->device);
|
|
|
496 return_ac->ac_left_speaker_gain = desc->ac_left_speaker_gain;
|
|
|
497 return_ac->ac_right_speaker_gain = desc->ac_right_speaker_gain;
|
|
|
498 params[0] = AL_OUTPUT_RATE;
|
|
|
499 params[1] = desc->ac_output_rate;
|
|
|
500 ALsetparams (desc->ac_device, params, 2);
|
|
|
501 return_ac->ac_output_rate = desc->ac_output_rate;
|
|
|
502 if (set_channels (config, desc->ac_nchan)==-1)
|
|
|
503 return -1;
|
|
|
504 return_ac->ac_nchan = desc->ac_nchan;
|
|
|
505 if (set_output_format (config, desc->ac_format)==-1)
|
|
|
506 return -1;
|
|
|
507 return_ac->ac_format = desc->ac_format;
|
|
|
508 ALsetqueuesize (config, (long) CHUNKSIZE);
|
|
|
509 return_ac->ac_port = ALopenport("XEmacs audio output", "w", config);
|
|
|
510 ALfreeconfig (config);
|
|
|
511 if (return_ac->ac_port==0)
|
|
|
512 {
|
|
563
|
513 report_sound_error ("Opening audio output port", Qunbound);
|
|
428
|
514 return -1;
|
|
|
515 }
|
|
|
516 return 0;
|
|
|
517 }
|
|
|
518
|
|
|
519 static int
|
|
|
520 set_channels (ALconfig config, unsigned int nchan)
|
|
|
521 {
|
|
|
522 switch (nchan)
|
|
|
523 {
|
|
|
524 case 1: ALsetchannels (config, AL_MONO); break;
|
|
|
525 #if HAVE_STEREO
|
|
|
526 case 2: ALsetchannels (config, AL_STEREO); break;
|
|
|
527 #endif /* HAVE_STEREO */
|
|
|
528 default:
|
|
563
|
529 report_sound_error ("Unsupported channel count",
|
|
|
530 make_int (nchan));
|
|
428
|
531 return -1;
|
|
|
532 }
|
|
|
533 return 0;
|
|
|
534 }
|
|
|
535
|
|
|
536 static int
|
|
|
537 set_output_format (ALconfig config, AudioFormat format)
|
|
|
538 {
|
|
|
539 long samplesize;
|
|
|
540 long old_samplesize;
|
|
|
541
|
|
|
542 switch (format)
|
|
|
543 {
|
|
|
544 #if HAVE_MULAW_8
|
|
|
545 case AFmulaw8:
|
|
|
546 #endif
|
|
|
547 #if HAVE_LINEAR
|
|
|
548 case AFlinear16:
|
|
|
549 #endif
|
|
|
550 #if HAVE_MULAW_8 || HAVE_LINEAR
|
|
|
551 samplesize = AL_SAMPLE_16;
|
|
|
552 break;
|
|
|
553 #endif
|
|
|
554 #if HAVE_LINEAR
|
|
|
555 case AFlinear8:
|
|
|
556 samplesize = AL_SAMPLE_8;
|
|
|
557 break;
|
|
|
558 case AFlinear24:
|
|
|
559 #if HAVE_LINEAR_32
|
|
|
560 case AFlinear32:
|
|
|
561 samplesize = AL_SAMPLE_24;
|
|
|
562 break;
|
|
|
563 #endif
|
|
|
564 #endif
|
|
|
565 default:
|
|
563
|
566 report_sound_error ("Unsupported audio format", make_int (format));
|
|
428
|
567 return -1;
|
|
|
568 }
|
|
|
569 old_samplesize = ALgetwidth (config);
|
|
|
570 if (old_samplesize==samplesize)
|
|
|
571 return 0;
|
|
|
572 ALsetwidth (config, samplesize);
|
|
|
573 return 1;
|
|
|
574 }
|
|
|
575
|
|
|
576 static void
|
|
|
577 adjust_audio_volume (AudioDevice device)
|
|
|
578 {
|
|
|
579 long params[4];
|
|
|
580 params[0] = AL_LEFT_SPEAKER_GAIN;
|
|
|
581 params[1] = device->left_speaker_gain;
|
|
|
582 params[2] = AL_RIGHT_SPEAKER_GAIN;
|
|
|
583 params[3] = device->right_speaker_gain;
|
|
|
584 ALsetparams (device->device, params, 4);
|
|
|
585 }
|
|
|
586
|
|
|
587 static void
|
|
|
588 get_current_volumes (AudioDevice device)
|
|
|
589 {
|
|
|
590 long params[4];
|
|
|
591 params[0] = AL_LEFT_SPEAKER_GAIN;
|
|
|
592 params[2] = AL_RIGHT_SPEAKER_GAIN;
|
|
|
593 ALgetparams (device->device, params, 4);
|
|
|
594 device->left_speaker_gain = params[1];
|
|
|
595 device->right_speaker_gain = params[3];
|
|
|
596 }
|
|
|
597
|
|
|
598 #if HAVE_SND_FILES
|
|
|
599
|
|
|
600 /* Parsing .snd (NeXT/Sun) headers */
|
|
|
601
|
|
|
602 typedef struct
|
|
|
603 {
|
|
|
604 int magic;
|
|
|
605 int dataLocation;
|
|
|
606 int dataSize;
|
|
|
607 int dataFormat;
|
|
|
608 int samplingRate;
|
|
|
609 int channelCount;
|
|
563
|
610 Char_Binary info[4];
|
|
428
|
611 }
|
|
|
612 SNDSoundStruct;
|
|
|
613 #define SOUND_TO_HOST_INT(x) ntohl(x)
|
|
|
614
|
|
|
615 typedef enum
|
|
|
616 {
|
|
|
617 SND_FORMAT_FORMAT_UNSPECIFIED,
|
|
|
618 SND_FORMAT_MULAW_8,
|
|
|
619 SND_FORMAT_LINEAR_8,
|
|
|
620 SND_FORMAT_LINEAR_16,
|
|
|
621 SND_FORMAT_LINEAR_24,
|
|
|
622 SND_FORMAT_LINEAR_32,
|
|
|
623 SND_FORMAT_FLOAT,
|
|
|
624 SND_FORMAT_DOUBLE,
|
|
|
625 SND_FORMAT_INDIRECT,
|
|
|
626 SND_FORMAT_NESTED,
|
|
|
627 SND_FORMAT_DSP_CODE,
|
|
|
628 SND_FORMAT_DSP_DATA_8,
|
|
|
629 SND_FORMAT_DSP_DATA_16,
|
|
|
630 SND_FORMAT_DSP_DATA_24,
|
|
|
631 SND_FORMAT_DSP_DATA_32,
|
|
|
632 SND_FORMAT_DSP_unknown_15,
|
|
|
633 SND_FORMAT_DISPLAY,
|
|
|
634 SND_FORMAT_MULAW_SQUELCH,
|
|
|
635 SND_FORMAT_EMPHASIZED,
|
|
|
636 SND_FORMAT_COMPRESSED,
|
|
|
637 SND_FORMAT_COMPRESSED_EMPHASIZED,
|
|
|
638 SND_FORMAT_DSP_COMMANDS,
|
|
|
639 SND_FORMAT_DSP_COMMANDS_SAMPLES
|
|
|
640 }
|
|
|
641 SNDFormatCode;
|
|
|
642
|
|
|
643 static int
|
|
|
644 parse_snd_header (void *header, long length, AudioContext desc)
|
|
|
645 {
|
|
|
646 #define hp ((SNDSoundStruct *) (header))
|
|
|
647 long limit;
|
|
|
648
|
|
|
649 #if HAVE_LINEAR
|
|
|
650 desc->ac_write_chunk_function = write_linear_chunk;
|
|
|
651 #endif
|
|
|
652 switch ((SNDFormatCode) SOUND_TO_HOST_INT (hp->dataFormat))
|
|
|
653 {
|
|
|
654 #if HAVE_MULAW_8
|
|
|
655 case SND_FORMAT_MULAW_8:
|
|
|
656 desc->ac_format = AFmulaw8;
|
|
|
657 desc->ac_write_chunk_function = write_mulaw_8_chunk;
|
|
|
658 break;
|
|
|
659 #endif
|
|
|
660 #if HAVE_LINEAR
|
|
|
661 case SND_FORMAT_LINEAR_8:
|
|
|
662 desc->ac_format = AFlinear8;
|
|
|
663 break;
|
|
|
664 case SND_FORMAT_LINEAR_16:
|
|
|
665 desc->ac_format = AFlinear16;
|
|
|
666 break;
|
|
|
667 case SND_FORMAT_LINEAR_24:
|
|
|
668 desc->ac_format = AFlinear24;
|
|
|
669 break;
|
|
|
670 #endif
|
|
|
671 #if HAVE_LINEAR_32
|
|
|
672 case SND_FORMAT_LINEAR_32:
|
|
|
673 desc->ac_format = AFlinear32;
|
|
|
674 desc->ac_write_chunk_function = write_linear_32_chunk;
|
|
|
675 break;
|
|
|
676 #endif
|
|
|
677 default:
|
|
|
678 desc->ac_format = AFunknown;
|
|
|
679 }
|
|
|
680 desc->ac_output_rate = SOUND_TO_HOST_INT (hp->samplingRate);
|
|
|
681 desc->ac_nchan = SOUND_TO_HOST_INT (hp->channelCount);
|
|
563
|
682 desc->ac_data = (Char_Binary *) header + SOUND_TO_HOST_INT (hp->dataLocation);
|
|
|
683 limit = (Char_Binary *) header + length - (Char_Binary *) desc->ac_data;
|
|
428
|
684 desc->ac_size = SOUND_TO_HOST_INT (hp->dataSize);
|
|
|
685 if (desc->ac_size > limit) desc->ac_size = limit;
|
|
|
686 return 0;
|
|
|
687 #undef hp
|
|
|
688 }
|
|
|
689 #endif /* HAVE_SND_FILES */
|
|
|
690
|
