1 | /* |
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2 | * Licensed to the Apache Software Foundation (ASF) under one |
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3 | * or more contributor license agreements. See the NOTICE file |
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4 | * distributed with this work for additional information |
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5 | * regarding copyright ownership. The ASF licenses this file |
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6 | * to you under the Apache License, Version 2.0 (the |
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7 | * "License"); you may not use this file except in compliance |
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8 | * with the License. You may obtain a copy of the License at |
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9 | * |
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10 | * http://www.apache.org/licenses/LICENSE-2.0 |
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11 | * |
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12 | * Unless required by applicable law or agreed to in writing, |
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13 | * software distributed under the License is distributed on an |
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14 | * "AS IS" BASIS, WITHOUT WARRANTIES OR CONDITIONS OF ANY |
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15 | * KIND, either express or implied. See the License for the |
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16 | * specific language governing permissions and limitations |
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17 | * under the License. |
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18 | */ |
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19 | |
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20 | var Guacamole = Guacamole || {}; |
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21 | |
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22 | /** |
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23 | * Abstract audio recorder which streams arbitrary audio data to an underlying |
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24 | * Guacamole.OutputStream. It is up to implementations of this class to provide |
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25 | * some means of handling this Guacamole.OutputStream. Data produced by the |
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26 | * recorder is to be sent along the provided stream immediately. |
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27 | * |
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28 | * @constructor |
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29 | */ |
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30 | Guacamole.AudioRecorder = function AudioRecorder() { |
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31 | |
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32 | /** |
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33 | * Callback which is invoked when the audio recording process has stopped |
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34 | * and the underlying Guacamole stream has been closed normally. Audio will |
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35 | * only resume recording if a new Guacamole.AudioRecorder is started. This |
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36 | * Guacamole.AudioRecorder instance MAY NOT be reused. |
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37 | * |
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38 | * @event |
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39 | */ |
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40 | this.onclose = null; |
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41 | |
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42 | /** |
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43 | * Callback which is invoked when the audio recording process cannot |
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44 | * continue due to an error, if it has started at all. The underlying |
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45 | * Guacamole stream is automatically closed. Future attempts to record |
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46 | * audio should not be made, and this Guacamole.AudioRecorder instance |
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47 | * MAY NOT be reused. |
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48 | * |
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49 | * @event |
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50 | */ |
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51 | this.onerror = null; |
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52 | |
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53 | }; |
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54 | |
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55 | /** |
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56 | * Determines whether the given mimetype is supported by any built-in |
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57 | * implementation of Guacamole.AudioRecorder, and thus will be properly handled |
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58 | * by Guacamole.AudioRecorder.getInstance(). |
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59 | * |
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60 | * @param {String} mimetype |
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61 | * The mimetype to check. |
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62 | * |
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63 | * @returns {Boolean} |
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64 | * true if the given mimetype is supported by any built-in |
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65 | * Guacamole.AudioRecorder, false otherwise. |
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66 | */ |
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67 | Guacamole.AudioRecorder.isSupportedType = function isSupportedType(mimetype) { |
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68 | |
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69 | return Guacamole.RawAudioRecorder.isSupportedType(mimetype); |
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70 | |
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71 | }; |
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72 | |
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73 | /** |
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74 | * Returns a list of all mimetypes supported by any built-in |
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75 | * Guacamole.AudioRecorder, in rough order of priority. Beware that only the |
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76 | * core mimetypes themselves will be listed. Any mimetype parameters, even |
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77 | * required ones, will not be included in the list. For example, "audio/L8" is |
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78 | * a supported raw audio mimetype that is supported, but it is invalid without |
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79 | * additional parameters. Something like "audio/L8;rate=44100" would be valid, |
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80 | * however (see https://tools.ietf.org/html/rfc4856). |
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81 | * |
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82 | * @returns {String[]} |
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83 | * A list of all mimetypes supported by any built-in |
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84 | * Guacamole.AudioRecorder, excluding any parameters. |
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85 | */ |
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86 | Guacamole.AudioRecorder.getSupportedTypes = function getSupportedTypes() { |
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87 | |
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88 | return Guacamole.RawAudioRecorder.getSupportedTypes(); |
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89 | |
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90 | }; |
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91 | |
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92 | /** |
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93 | * Returns an instance of Guacamole.AudioRecorder providing support for the |
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94 | * given audio format. If support for the given audio format is not available, |
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95 | * null is returned. |
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96 | * |
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97 | * @param {Guacamole.OutputStream} stream |
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98 | * The Guacamole.OutputStream to send audio data through. |
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99 | * |
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100 | * @param {String} mimetype |
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101 | * The mimetype of the audio data to be sent along the provided stream. |
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102 | * |
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103 | * @return {Guacamole.AudioRecorder} |
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104 | * A Guacamole.AudioRecorder instance supporting the given mimetype and |
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105 | * writing to the given stream, or null if support for the given mimetype |
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106 | * is absent. |
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107 | */ |
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108 | Guacamole.AudioRecorder.getInstance = function getInstance(stream, mimetype) { |
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109 | |
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110 | // Use raw audio recorder if possible |
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111 | if (Guacamole.RawAudioRecorder.isSupportedType(mimetype)) |
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112 | return new Guacamole.RawAudioRecorder(stream, mimetype); |
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113 | |
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114 | // No support for given mimetype |
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115 | return null; |
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116 | |
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117 | }; |
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118 | |
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119 | /** |
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120 | * Implementation of Guacamole.AudioRecorder providing support for raw PCM |
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121 | * format audio. This recorder relies only on the Web Audio API and does not |
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122 | * require any browser-level support for its audio formats. |
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123 | * |
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124 | * @constructor |
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125 | * @augments Guacamole.AudioRecorder |
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126 | * @param {Guacamole.OutputStream} stream |
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127 | * The Guacamole.OutputStream to write audio data to. |
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128 | * |
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129 | * @param {String} mimetype |
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130 | * The mimetype of the audio data to send along the provided stream, which |
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131 | * must be a "audio/L8" or "audio/L16" mimetype with necessary parameters, |
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132 | * such as: "audio/L16;rate=44100,channels=2". |
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133 | */ |
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134 | Guacamole.RawAudioRecorder = function RawAudioRecorder(stream, mimetype) { |
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135 | |
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136 | /** |
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137 | * Reference to this RawAudioRecorder. |
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138 | * |
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139 | * @private |
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140 | * @type {Guacamole.RawAudioRecorder} |
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141 | */ |
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142 | var recorder = this; |
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143 | |
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144 | /** |
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145 | * The size of audio buffer to request from the Web Audio API when |
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146 | * recording or processing audio, in sample-frames. This must be a power of |
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147 | * two between 256 and 16384 inclusive, as required by |
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148 | * AudioContext.createScriptProcessor(). |
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149 | * |
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150 | * @private |
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151 | * @constant |
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152 | * @type {Number} |
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153 | */ |
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154 | var BUFFER_SIZE = 2048; |
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155 | |
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156 | /** |
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157 | * The window size to use when applying Lanczos interpolation, commonly |
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158 | * denoted by the variable "a". |
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159 | * See: https://en.wikipedia.org/wiki/Lanczos_resampling |
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160 | * |
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161 | * @private |
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162 | * @contant |
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163 | * @type Number |
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164 | */ |
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165 | var LANCZOS_WINDOW_SIZE = 3; |
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166 | |
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167 | /** |
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168 | * The format of audio this recorder will encode. |
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169 | * |
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170 | * @private |
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171 | * @type {Guacamole.RawAudioFormat} |
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172 | */ |
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173 | var format = Guacamole.RawAudioFormat.parse(mimetype); |
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174 | |
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175 | /** |
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176 | * An instance of a Web Audio API AudioContext object, or null if the |
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177 | * Web Audio API is not supported. |
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178 | * |
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179 | * @private |
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180 | * @type {AudioContext} |
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181 | */ |
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182 | var context = Guacamole.AudioContextFactory.getAudioContext(); |
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183 | |
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184 | /** |
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185 | * A function which directly invokes the browser's implementation of |
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186 | * navigator.getUserMedia() with all provided parameters. |
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187 | * |
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188 | * @type Function |
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189 | */ |
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190 | var getUserMedia = (navigator.getUserMedia |
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191 | || navigator.webkitGetUserMedia |
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192 | || navigator.mozGetUserMedia |
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193 | || navigator.msGetUserMedia).bind(navigator); |
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194 | |
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195 | /** |
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196 | * Guacamole.ArrayBufferWriter wrapped around the audio output stream |
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197 | * provided when this Guacamole.RawAudioRecorder was created. |
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198 | * |
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199 | * @private |
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200 | * @type {Guacamole.ArrayBufferWriter} |
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201 | */ |
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202 | var writer = new Guacamole.ArrayBufferWriter(stream); |
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203 | |
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204 | /** |
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205 | * The type of typed array that will be used to represent each audio packet |
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206 | * internally. This will be either Int8Array or Int16Array, depending on |
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207 | * whether the raw audio format is 8-bit or 16-bit. |
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208 | * |
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209 | * @private |
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210 | * @constructor |
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211 | */ |
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212 | var SampleArray = (format.bytesPerSample === 1) ? window.Int8Array : window.Int16Array; |
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213 | |
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214 | /** |
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215 | * The maximum absolute value of any sample within a raw audio packet sent |
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216 | * by this audio recorder. This depends only on the size of each sample, |
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217 | * and will be 128 for 8-bit audio and 32768 for 16-bit audio. |
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218 | * |
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219 | * @private |
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220 | * @type {Number} |
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221 | */ |
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222 | var maxSampleValue = (format.bytesPerSample === 1) ? 128 : 32768; |
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223 | |
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224 | /** |
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225 | * The total number of audio samples read from the local audio input device |
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226 | * over the life of this audio recorder. |
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227 | * |
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228 | * @private |
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229 | * @type {Number} |
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230 | */ |
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231 | var readSamples = 0; |
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232 | |
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233 | /** |
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234 | * The total number of audio samples written to the underlying Guacamole |
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235 | * connection over the life of this audio recorder. |
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236 | * |
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237 | * @private |
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238 | * @type {Number} |
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239 | */ |
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240 | var writtenSamples = 0; |
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241 | |
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242 | /** |
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243 | * The audio stream provided by the browser, if allowed. If no stream has |
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244 | * yet been received, this will be null. |
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245 | * |
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246 | * @type MediaStream |
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247 | */ |
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248 | var mediaStream = null; |
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249 | |
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250 | /** |
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251 | * The source node providing access to the local audio input device. |
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252 | * |
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253 | * @private |
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254 | * @type {MediaStreamAudioSourceNode} |
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255 | */ |
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256 | var source = null; |
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257 | |
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258 | /** |
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259 | * The script processing node which receives audio input from the media |
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260 | * stream source node as individual audio buffers. |
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261 | * |
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262 | * @private |
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263 | * @type {ScriptProcessorNode} |
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264 | */ |
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265 | var processor = null; |
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266 | |
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267 | /** |
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268 | * The normalized sinc function. The normalized sinc function is defined as |
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269 | * 1 for x=0 and sin(PI * x) / (PI * x) for all other values of x. |
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270 | * |
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271 | * See: https://en.wikipedia.org/wiki/Sinc_function |
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272 | * |
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273 | * @private |
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274 | * @param {Number} x |
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275 | * The point at which the normalized sinc function should be computed. |
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276 | * |
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277 | * @returns {Number} |
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278 | * The value of the normalized sinc function at x. |
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279 | */ |
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280 | var sinc = function sinc(x) { |
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281 | |
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282 | // The value of sinc(0) is defined as 1 |
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283 | if (x === 0) |
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284 | return 1; |
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285 | |
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286 | // Otherwise, normlized sinc(x) is sin(PI * x) / (PI * x) |
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287 | var piX = Math.PI * x; |
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288 | return Math.sin(piX) / piX; |
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289 | |
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290 | }; |
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291 | |
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292 | /** |
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293 | * Calculates the value of the Lanczos kernal at point x for a given window |
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294 | * size. See: https://en.wikipedia.org/wiki/Lanczos_resampling |
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295 | * |
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296 | * @private |
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297 | * @param {Number} x |
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298 | * The point at which the value of the Lanczos kernel should be |
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299 | * computed. |
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300 | * |
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301 | * @param {Number} a |
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302 | * The window size to use for the Lanczos kernel. |
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303 | * |
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304 | * @returns {Number} |
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305 | * The value of the Lanczos kernel at the given point for the given |
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306 | * window size. |
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307 | */ |
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308 | var lanczos = function lanczos(x, a) { |
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309 | |
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310 | // Lanczos is sinc(x) * sinc(x / a) for -a < x < a ... |
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311 | if (-a < x && x < a) |
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312 | return sinc(x) * sinc(x / a); |
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313 | |
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314 | // ... and 0 otherwise |
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315 | return 0; |
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316 | |
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317 | }; |
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318 | |
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319 | /** |
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320 | * Determines the value of the waveform represented by the audio data at |
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321 | * the given location. If the value cannot be determined exactly as it does |
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322 | * not correspond to an exact sample within the audio data, the value will |
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323 | * be derived through interpolating nearby samples. |
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324 | * |
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325 | * @private |
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326 | * @param {Float32Array} audioData |
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327 | * An array of audio data, as returned by AudioBuffer.getChannelData(). |
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328 | * |
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329 | * @param {Number} t |
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330 | * The relative location within the waveform from which the value |
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331 | * should be retrieved, represented as a floating point number between |
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332 | * 0 and 1 inclusive, where 0 represents the earliest point in time and |
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333 | * 1 represents the latest. |
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334 | * |
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335 | * @returns {Number} |
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336 | * The value of the waveform at the given location. |
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337 | */ |
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338 | var interpolateSample = function getValueAt(audioData, t) { |
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339 | |
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340 | // Convert [0, 1] range to [0, audioData.length - 1] |
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341 | var index = (audioData.length - 1) * t; |
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342 | |
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343 | // Determine the start and end points for the summation used by the |
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344 | // Lanczos interpolation algorithm (see: https://en.wikipedia.org/wiki/Lanczos_resampling) |
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345 | var start = Math.floor(index) - LANCZOS_WINDOW_SIZE + 1; |
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346 | var end = Math.floor(index) + LANCZOS_WINDOW_SIZE; |
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347 | |
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348 | // Calculate the value of the Lanczos interpolation function for the |
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349 | // required range |
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350 | var sum = 0; |
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351 | for (var i = start; i <= end; i++) { |
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352 | sum += (audioData[i] || 0) * lanczos(index - i, LANCZOS_WINDOW_SIZE); |
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353 | } |
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354 | |
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355 | return sum; |
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356 | |
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357 | }; |
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358 | |
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359 | /** |
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360 | * Converts the given AudioBuffer into an audio packet, ready for streaming |
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361 | * along the underlying output stream. Unlike the raw audio packets used by |
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362 | * this audio recorder, AudioBuffers require floating point samples and are |
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363 | * split into isolated planes of channel-specific data. |
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364 | * |
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365 | * @private |
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366 | * @param {AudioBuffer} audioBuffer |
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367 | * The Web Audio API AudioBuffer that should be converted to a raw |
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368 | * audio packet. |
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369 | * |
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370 | * @returns {SampleArray} |
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371 | * A new raw audio packet containing the audio data from the provided |
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372 | * AudioBuffer. |
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373 | */ |
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374 | var toSampleArray = function toSampleArray(audioBuffer) { |
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375 | |
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376 | // Track overall amount of data read |
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377 | var inSamples = audioBuffer.length; |
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378 | readSamples += inSamples; |
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379 | |
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380 | // Calculate the total number of samples that should be written as of |
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381 | // the audio data just received and adjust the size of the output |
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382 | // packet accordingly |
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383 | var expectedWrittenSamples = Math.round(readSamples * format.rate / audioBuffer.sampleRate); |
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384 | var outSamples = expectedWrittenSamples - writtenSamples; |
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385 | |
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386 | // Update number of samples written |
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387 | writtenSamples += outSamples; |
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388 | |
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389 | // Get array for raw PCM storage |
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390 | var data = new SampleArray(outSamples * format.channels); |
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391 | |
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392 | // Convert each channel |
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393 | for (var channel = 0; channel < format.channels; channel++) { |
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394 | |
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395 | var audioData = audioBuffer.getChannelData(channel); |
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396 | |
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397 | // Fill array with data from audio buffer channel |
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398 | var offset = channel; |
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399 | for (var i = 0; i < outSamples; i++) { |
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400 | data[offset] = interpolateSample(audioData, i / (outSamples - 1)) * maxSampleValue; |
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401 | offset += format.channels; |
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402 | } |
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403 | |
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404 | } |
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405 | |
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406 | return data; |
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407 | |
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408 | }; |
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409 | |
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410 | /** |
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411 | * Requests access to the user's microphone and begins capturing audio. All |
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412 | * received audio data is resampled as necessary and forwarded to the |
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413 | * Guacamole stream underlying this Guacamole.RawAudioRecorder. This |
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414 | * function must be invoked ONLY ONCE per instance of |
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415 | * Guacamole.RawAudioRecorder. |
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416 | * |
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417 | * @private |
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418 | */ |
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419 | var beginAudioCapture = function beginAudioCapture() { |
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420 | |
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421 | // Attempt to retrieve an audio input stream from the browser |
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422 | getUserMedia({ 'audio' : true }, function streamReceived(stream) { |
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423 | |
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424 | // Create processing node which receives appropriately-sized audio buffers |
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425 | processor = context.createScriptProcessor(BUFFER_SIZE, format.channels, format.channels); |
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426 | processor.connect(context.destination); |
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427 | |
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428 | // Send blobs when audio buffers are received |
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429 | processor.onaudioprocess = function processAudio(e) { |
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430 | writer.sendData(toSampleArray(e.inputBuffer).buffer); |
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431 | }; |
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432 | |
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433 | // Connect processing node to user's audio input source |
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434 | source = context.createMediaStreamSource(stream); |
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435 | source.connect(processor); |
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436 | |
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437 | // Save stream for later cleanup |
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438 | mediaStream = stream; |
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439 | |
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440 | }, function streamDenied() { |
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441 | |
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442 | // Simply end stream if audio access is not allowed |
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443 | writer.sendEnd(); |
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444 | |
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445 | // Notify of closure |
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446 | if (recorder.onerror) |
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447 | recorder.onerror(); |
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448 | |
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449 | }); |
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450 | |
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451 | }; |
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452 | |
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453 | /** |
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454 | * Stops capturing audio, if the capture has started, freeing all associated |
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455 | * resources. If the capture has not started, this function simply ends the |
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456 | * underlying Guacamole stream. |
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457 | * |
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458 | * @private |
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459 | */ |
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460 | var stopAudioCapture = function stopAudioCapture() { |
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461 | |
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462 | // Disconnect media source node from script processor |
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463 | if (source) |
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464 | source.disconnect(); |
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465 | |
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466 | // Disconnect associated script processor node |
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467 | if (processor) |
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468 | processor.disconnect(); |
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469 | |
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470 | // Stop capture |
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471 | if (mediaStream) { |
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472 | var tracks = mediaStream.getTracks(); |
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473 | for (var i = 0; i < tracks.length; i++) |
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474 | tracks[i].stop(); |
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475 | } |
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476 | |
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477 | // Remove references to now-unneeded components |
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478 | processor = null; |
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479 | source = null; |
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480 | mediaStream = null; |
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481 | |
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482 | // End stream |
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483 | writer.sendEnd(); |
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484 | |
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485 | }; |
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486 | |
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487 | // Once audio stream is successfully open, request and begin reading audio |
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488 | writer.onack = function audioStreamAcknowledged(status) { |
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489 | |
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490 | // Begin capture if successful response and not yet started |
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491 | if (status.code === Guacamole.Status.Code.SUCCESS && !mediaStream) |
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492 | beginAudioCapture(); |
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493 | |
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494 | // Otherwise stop capture and cease handling any further acks |
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495 | else { |
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496 | |
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497 | // Stop capturing audio |
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498 | stopAudioCapture(); |
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499 | writer.onack = null; |
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500 | |
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501 | // Notify if stream has closed normally |
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502 | if (status.code === Guacamole.Status.Code.RESOURCE_CLOSED) { |
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503 | if (recorder.onclose) |
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504 | recorder.onclose(); |
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505 | } |
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506 | |
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507 | // Otherwise notify of closure due to error |
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508 | else { |
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509 | if (recorder.onerror) |
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510 | recorder.onerror(); |
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511 | } |
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512 | |
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513 | } |
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514 | |
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515 | }; |
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516 | |
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517 | }; |
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518 | |
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519 | Guacamole.RawAudioRecorder.prototype = new Guacamole.AudioRecorder(); |
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520 | |
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521 | /** |
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522 | * Determines whether the given mimetype is supported by |
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523 | * Guacamole.RawAudioRecorder. |
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524 | * |
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525 | * @param {String} mimetype |
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526 | * The mimetype to check. |
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527 | * |
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528 | * @returns {Boolean} |
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529 | * true if the given mimetype is supported by Guacamole.RawAudioRecorder, |
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530 | * false otherwise. |
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531 | */ |
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532 | Guacamole.RawAudioRecorder.isSupportedType = function isSupportedType(mimetype) { |
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533 | |
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534 | // No supported types if no Web Audio API |
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535 | if (!Guacamole.AudioContextFactory.getAudioContext()) |
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536 | return false; |
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537 | |
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538 | return Guacamole.RawAudioFormat.parse(mimetype) !== null; |
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539 | |
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540 | }; |
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541 | |
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542 | /** |
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543 | * Returns a list of all mimetypes supported by Guacamole.RawAudioRecorder. Only |
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544 | * the core mimetypes themselves will be listed. Any mimetype parameters, even |
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545 | * required ones, will not be included in the list. For example, "audio/L8" is |
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546 | * a raw audio mimetype that may be supported, but it is invalid without |
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547 | * additional parameters. Something like "audio/L8;rate=44100" would be valid, |
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548 | * however (see https://tools.ietf.org/html/rfc4856). |
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549 | * |
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550 | * @returns {String[]} |
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551 | * A list of all mimetypes supported by Guacamole.RawAudioRecorder, |
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552 | * excluding any parameters. If the necessary JavaScript APIs for recording |
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553 | * raw audio are absent, this list will be empty. |
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554 | */ |
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555 | Guacamole.RawAudioRecorder.getSupportedTypes = function getSupportedTypes() { |
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556 | |
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557 | // No supported types if no Web Audio API |
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558 | if (!Guacamole.AudioContextFactory.getAudioContext()) |
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559 | return []; |
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560 | |
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561 | // We support 8-bit and 16-bit raw PCM |
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562 | return [ |
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563 | 'audio/L8', |
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564 | 'audio/L16' |
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565 | ]; |
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566 | |
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567 | }; |
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