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