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