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