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4807 lines
229 KiB
4807 lines
229 KiB
<!DOCTYPE html>
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<html>
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<meta charset="utf-8">
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<meta name="viewport" content="width=device-width, initial-scale=1, maximum-scale=1, user-scalable=0"/>
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<meta name="apple-mobile-web-app-capable" content="yes">
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<meta name="mobile-web-app-capable" content="yes">
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<!--
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Modified from original Node-Red source, for audio system visualization
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Copyright 2013 IBM Corp.
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Licensed under the Apache License, Version 2.0 (the "License");
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you may not use this file except in compliance with the License.
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You may obtain a copy of the License at
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http://www.apache.org/licenses/LICENSE-2.0
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Unless required by applicable law or agreed to in writing, software
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distributed under the License is distributed on an "AS IS" BASIS,
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WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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See the License for the specific language governing permissions and
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limitations under the License.
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*** NODES FOR OPENAUDIO_ARDUINOLIBRARY 1 March 2021 RSL ***
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-->
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<head>
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<title>Audio System Design Tool for Open Audio F32 Library for Teensy</title>
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<link href="bootstrap/css/bootstrap.min.css" rel="stylesheet" media="screen">
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<link href="jquery/css/smoothness/jquery-ui-1.10.3.custom.min.css" rel="stylesheet" media="screen">
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<link rel="stylesheet" type="text/css" href="orion/built-editor.css"/>
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<link rel="stylesheet" type="text/css" href="font-awesome/css/font-awesome.min.css"/>
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<link rel="stylesheet" href="style.css">
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<style>
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table.doc {border-spacing:3px; border-collapse:separate; font-size: 80%}
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tr.top {background-color:#C0C0C0}
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tr.odd {background-color:#F0F0F0}
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tr.even {background-color:#E0E0E0}
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p.func {padding-bottom:0; margin:0px}
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p.desc {padding-left:2em; margin:0px; padding-top:0.2em; padding-bottom:0.8em; font-size:0.75em}
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p.exam {padding-left:2em; text-indent:-1.2em; margin:0px; padding-top:0; padding-bottom:0.5em; font-size:0.75em; font-weight:bold}
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pre.desc {padding-left:3em; margin:0px; padding-top:0em; padding-bottom:0.8em; font-size:0.75em;
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background-color:#FFFFFF; border:0px; line-height:100%;
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}
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span.indent {padding-left:2em}
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span.literal {color: #006699}
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span.comment {color: #777755}
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span.keyword {color: #cc6600}
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span.function {color: #996600}
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span.mainfunction {color: #993300; font-weight: bolder}
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</style>
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</head>
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<body spellcheck="false">
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<div class="navbar navbar-inverse navbar-fixed-top">
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<div class="navbar-inner">
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<div class="container-fluid">
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<span class="brand">Audio Design Tool for OpenAudio F32 Library for Teensy</a></span>
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<div class="btn-group pull-right">
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<a class="btn dropdown-toggle" data-toggle="dropdown" href="#"><i class="icon-align-justify"></i> <span class="caret"></span></a>
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<li><a id="btn-sidebar" tabindex="-1" href="#"><i class="icon-ok pull-right"></i><i class="icon-list-alt"></i> Sidebar</a></li>
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<li class="divider"></li>
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<!-- <li><a id="btn-node-status" tabindex="-1" href="#"><i class="icon-ok pull-right"></i><i class="icon-info-sign"></i> Node Status</a></li>
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<li class="divider"></li>
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-->
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<!--
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<li class="dropdown-submenu pull-left"><a tabindex="-1" href="#"><i class="icon-edit"></i> Import from...</a>
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<ul class="dropdown-menu">
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<li><a id="btn-import" tabindex="-1" href="#"><i class="icon-edit"></i> Clipboard...</a></li>
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<li id="flow-menu-parent" class="dropdown-submenu pull-left">
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<a tabindex="-1" href="#"><i class="icon-book"></i> Library</a>
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<ul class="dropdown-menu"></ul>
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</li>
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</ul>
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</li>
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<li id="li-menu-export" class="dropdown-submenu disabled pull-left"><a tabindex="-1" href="#"><i class="icon-share"></i> Export to...</a>
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<ul class="dropdown-menu">
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<li id="li-menu-export-clipboard" class="disabled"><a id="btn-export-clipboard" tabindex="-1" href="#"><i class="icon-share"></i> Clipboard...</a></li>
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<li id="li-menu-export-library" class="disabled"><a id="btn-export-library" tabindex="-1" href="#"><i class="icon-book"></i> Library...</a></li>
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</ul>
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</li>
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<li class="divider"></li>
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-->
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<!--
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<li><a id="btn-config-nodes" tabindex="-1" href="#"><i class="icon-th-list"></i> Configuration nodes...</a></li>
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<li class="divider"></li>
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-->
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<!--
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<li class="dropdown-submenu pull-left"><a tabindex="-1" href="#"><i class="icon-th-large"></i> Workspaces</a>
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<ul id="workspace-menu-list" class="dropdown-menu">
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<li><a id="btn-workspace-add" tabindex="-1" href="#"><i class="icon-plus"></i> Add</a></li>
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<li><a id="btn-workspace-edit" tabindex="-1" href="#"><i class="icon-edit"></i> Rename</a></li>
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<li><a id="btn-workspace-delete" tabindex="-1" href="#"><i class="icon-minus"></i> Delete</a></li>
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<li class="divider"></li>
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</ul>
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</li>
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<li class="divider"></li>-->
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<li><a id="btn-keyboard-shortcuts" tabindex="-1" href="#"><i class="icon-question-sign"></i> Keyboard Shortcuts</a></li>
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<li><a id="btn-help" tabindex="-1" href="http://node-red.github.io/docs" target="_blank"><i class="icon-question-sign"></i> Help...</a></li>
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</ul>
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</div>
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<div class="btn-group pull-left">
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<a id="btn-deploy" class="btn action-deploy disabled" href="#"><i id="btn-icn-deploy" class="icon-upload"></i>Export</a>
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<a id="btn-import" class="btn action-import disabled" href="#"><i id="btn-icn-download" class="icon-download"></i>Import</a>
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</div>
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</div>
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</div>
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</div>
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<div id="main-container" class="sidebar-closed">
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<div id="palette">
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<img src="img/spin.svg" class="palette-spinner"/>
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<div id="palette-container" class="palette-scroll">
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</div>
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<div id="palette-search">
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<i class="icon-search"></i><input id="palette-search-input" type="text" placeholder="filter"><a href="#" id="palette-search-clear"><i class="icon-remove"></i></a></input>
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</div>
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</div><!-- /palette -->
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<div id="workspace">
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<ul id="workspace-tabs"></ul>
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<!--<div id="workspace-add-tab"><a id="btn-workspace-add-tab" href="#"><i class="icon-plus"></i></a></div>-->
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<div id="chart"></div>
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<div id="workspace-toolbar">
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<div class="btn-group">
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<a class="btn btn-small" href="#"><i class="icon-zoom-out"></i></a>
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<a class="btn btn-small" href="#"><i class="icon-th"></i></a>
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<a class="btn btn-small" href="#"><i class="icon-zoom-in"></i></a>
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</div>
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</div>
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</div>
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<div id="chart-zoom-controls">
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<div class="btn-group">
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<a class="btn btn-mini" id="btn-zoom-out" href="#"><i class="icon-zoom-out"></i></a>
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<a class="btn btn-mini" id="btn-zoom-zero" href="#"><i class="icon-th"></i></a>
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<a class="btn btn-mini" id="btn-zoom-in" href="#"><i class="icon-zoom-in"></i></a>
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</div>
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</div>
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<div id="sidebar">
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<ul id="sidebar-tabs"></ul>
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<div id="sidebar-content"></div>
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</div>
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<div id="sidebar-separator"></div>
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</div>
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<div id="notifications"></div>
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<div id="dropTarget"><div>Drop the flow here</div></div>
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<div id="dialog" class="hide"><form id="dialog-form" class="form-horizontal"></form></div>
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<div id="node-config-dialog" class="hide"><form id="dialog-config-form" class="form-horizontal"></form><div class="form-tips" id="node-config-dialog-user-count"></div></div>
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<div id="node-dialog-confirm-deploy" class="hide">
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<form class="form-horizontal">
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<div id="node-dialog-confirm-deploy-config" style="text-align: center; padding-top: 30px;">
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Some of the nodes are not properly configured. Are you sure you want to deploy?
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</div>
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<div id="node-dialog-confirm-deploy-unknown" style="text-align: center; padding-top: 10px;">
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The workspace contains some unknown node types:
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<ul style="width: 300px; margin: auto; text-align: left;" id="node-dialog-confirm-deploy-unknown-list"></ul>
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Are you sure you want to deploy?
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</div>
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</form>
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</div>
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<div id="node-dialog-error-deploy" class="hide">
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<form class="form-horizontal">
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<div id="node-dialog-error-deploy-noio" style="text-align: center; padding-top: 10px;">
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<p>The workspace contains no input/output nodes!</p>
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<p>You need an input or an output to export the data!</p>
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<p>Without such a input/output function the exported
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code will not run properly!</p>
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</div>
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</form>
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</div>
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<div id="node-help" class="modal hide fade" tabindex="-1" role="dialog" aria-labelledby="node-help-label" aria-hidden="true">
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<div class="modal-header">
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<h5 id="node-help-label">Keyboard Shortcuts <span style="float: right;"><a href="http://node-red.github.io/docs" target="_blank">Open help in new window »</a></span></h5>
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</div>
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<div class="modal-body">
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<table>
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<tr>
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<td><span class="help-key">?</span></td><td>Help</td>
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<td><span class="help-key">Ctrl</span> <span class="help-key">a</span></td><td>Select all nodes</td>
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</tr>
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<tr>
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<td><span class="help-key">Ctrl</span> <span class="help-key">Space</span></td><td>Toggle sidebar</td>
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<td><span class="help-key">Shift</span> <span class="help-key">Click</span></td><td>Select all connected nodes</td>
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</tr>
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<tr>
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<td><span class="help-key">Ctrl</span> <span class="help-key">z</span></td><td>Undo</td>
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<td><span class="help-key">Ctrl</span> <span class="help-key">Click</span></td><td>Add/remove node from selection</td>
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</tr>
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<tr>
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<td></td><td></td>
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<td><span class="help-key">Delete</span></td><td>Delete selected nodes or link</td>
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</tr>
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<tr>
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<td><span class="help-key">Ctrl</span> <span class="help-key">x</span></td><td>Cut selected nodes</td>
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<td></td><td></td>
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</tr>
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<tr>
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<td><span class="help-key">Ctrl</span> <span class="help-key">c</span></td><td>Copy selected nodes</td>
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<td><span class="help-key">Ctrl</span> <span class="help-key">v</span></td><td>Paste nodes</td>
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</tr>
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<tr>
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<td><span class="help-key">Ctrl</span> <span class="help-key">i</span></td><td>Import nodes</td>
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<td><span class="help-key">Ctrl</span> <span class="help-key">e</span></td><td>Export selected nodes</td>
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</tr>
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<tr>
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<td colspan="2"></td>
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</tr>
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<tr>
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<td><span class="help-key">Ctrl</span> <span class="help-key">+</span></td><td>Zoom in</td>
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<td><span class="help-key">Ctrl</span> <span class="help-key">-</span></td><td>Zoom out</td>
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</tr>
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</table>
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</div>
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<div class="modal-footer">
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<button class="btn" data-dismiss="modal" aria-hidden="true">Close</button>
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</div>
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</div>
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<div id="node-dialog-library-save-confirm" class="hide">
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<form class="form-horizontal">
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<div style="text-align: center; padding-top: 30px;">
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A <span id="node-dialog-library-save-type"></span> called <span id="node-dialog-library-save-name"></span> already exists. Overwrite?
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</div>
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</form>
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</div>
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<div id="node-dialog-library-save" class="hide">
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<form class="form-horizontal">
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<div class="form-row">
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<label for="node-dialog-library-save-folder"><i class="icon-folder-open"></i> Folder</label>
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<input type="text" id="node-dialog-library-save-folder" placeholder="Folder">
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</div>
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<div class="form-row">
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<label for="node-dialog-library-save-filename"><i class="icon-file"></i> Filename</label>
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<input type="text" id="node-dialog-library-save-filename" placeholder="Filename">
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</div>
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</form>
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</div>
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<div id="node-dialog-library-lookup" class="hide">
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<form class="form-horizontal">
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<div class="form-row">
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<ul id="node-dialog-library-breadcrumbs" class="breadcrumb">
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<li class="active"><a href="#">Library</a></li>
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</ul>
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</div>
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<div class="form-row">
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<div style="vertical-align: top; display: inline-block; height: 100%; width: 30%; padding-right: 20px;">
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<div id="node-select-library" style="border: 1px solid #999; width: 100%; height: 100%; overflow:scroll;"><ul></ul></div>
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</div>
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<div style="vertical-align: top; display: inline-block;width: 65%; height: 100%;">
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<div style="height: 100%; width: 95%;" class="node-text-editor" id="node-select-library-text" ></div>
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</div>
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</div>
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</form>
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</div>
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<div id="node-dialog-rename-workspace" class="hide">
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<form class="form-horizontal">
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<div class="form-row">
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<label for="node-input-workspace-name" ><i class="icon-tag"></i> Name:</label>
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<input type="text" id="node-input-workspace-name">
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</div>
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</form>
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</div>
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<div id="node-dialog-delete-workspace" class="hide">
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<form class="form-horizontal">
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<div style="text-align: center; padding-top: 30px;">
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Are you sure you want to delete '<span id="node-dialog-delete-workspace-name"></span>'?
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</div>
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</form>
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</div>
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<script type="text/x-red" data-template-name="export-clipboard-dialog">
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<div class="form-row">
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<label for="node-input-export" style="display: block; width:100%;"><i class="icon-share"></i> Source Code:</label>
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<textarea readonly style="font-family: monospace; font-size: 12px; background:rgb(226, 229, 255); padding-left: 0.5em;" class="input-block-level" id="node-input-export" rows="12"></textarea>
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</div>
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<div class="form-tips">
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<a id="download-INO" target="_blank">Click to Download Zipped Code</a>
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</div>
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</script>
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<script type="text/x-red" data-template-name="export-library-dialog">
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<div class="form-row">
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<label for="node-input-filename" ><i class="icon-tag"></i> Filename:</label>
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<input type="text" id="node-input-filename" placeholder="Filename">
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</div>
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</script>
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<script type="text/x-red" data-template-name="import-dialog">
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<div class="form-row">
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<label for="node-input-import"><i class="icon-share"></i>Nodes:</label>
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<textarea style="font-family: monospace; font-size: 12px; background:rgb(226, 229, 255); padding-left: 0.5em;" class="input-block-level" id="node-input-import" rows="5" placeholder="Paste nodes here, or lookup in the library. When importing Arduino code, the whole flow will be replaced."></textarea>
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</div>
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<div class="form-tips">
|
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<label for="node-input-arduino" style="font-size: 13px; padding: 2px 0px 0px 4px;">
|
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<input style="margin-bottom: 4px; margin-right: 4px;" type="checkbox" id="node-input-arduino" checked="checked" class="input-block-level" />
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Import copied code from the Arduino IDE
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</label>
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</div>
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</script>
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<script src="jquery/js/jquery-1.9.1.js"></script>
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<script src="bootstrap/js/bootstrap.min.js"></script>
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<script src="jquery/js/jquery-ui-1.10.3.custom.min.js"></script>
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<script src="jquery/js/jquery.ui.touch-punch.min.js"></script>
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<script src="jszip/dist/jszip.min.js"></script>
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<script src="orion/built-editor.min.js"></script>
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<script src="red/d3/d3.v3.min.js"></script>
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<script src="red/main.js"></script>
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<script src="red/ui/state.js"></script>
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<script src="red/nodes.js"></script>
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<script src="red/storage.js"></script>
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<script src="red/history.js"></script>
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<script src="red/ui/keyboard.js"></script>
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<script src="red/ui/tabs.js"></script>
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<script src="red/ui/view.js"></script>
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<script src="red/ui/sidebar.js"></script>
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<script src="red/ui/palette.js"></script>
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<script src="red/ui/tab-info.js"></script>
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<script src="red/ui/tab-config.js"></script>
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<script src="red/ui/editor.js"></script>
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<script src="red/ui/library.js"></script>
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<script src="red/ui/notifications.js"></script>
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<script src="red/ui/touch/radialMenu.js"></script>
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<!--
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TODO: generate some or all of this automatically from the C++ source
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-->
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<!--
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TODO: add a field for maximum instance count
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-->
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<!--
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TODO: add a field for exclusive to other objects (not allowed if they're used)
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-->
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<!--
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TODO: add "parameters" fields, to replace the form html stuff
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-->
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<script type="text/x-red" data-container-name="NodeDefinitions">
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{"nodes":[
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{"type":"AudioAnalyzePhase_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"phaseDet","inputs":"1","output":"0","category":"analyze-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"1"}},
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{"type":"AudioAnalyzePeak_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"peak","inputs":1,"outputs":0,"category":"analyze-function","color":"#E6E0F8","icon":"arrow-in.png"}},
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{"type":"AudioAnalyzeRMS_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"rms","inputs":1,"outputs":0,"category":"analyze-function","color":"#E6E0F8","icon":"arrow-in.png"}},
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{"type":"AudioAnalyzeFFT1024_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"FFT1024","inputs":1,"outputs":0,"category":"analyze-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"0"}},
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{"type":"AudioAnalyzeFFT256_IQ_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"FFT256iq","inputs":2,"outputs":0,"category":"analyze-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"4"}},
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{"type":"AudioAnalyzeFFT1024_IQ_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"FFT1024iq","inputs":2,"outputs":0,"category":"analyze-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"4"}},
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|
{"type":"AudioAnalyzeFFT2048_IQ_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"FFT2048iq","inputs":2,"outputs":0,"category":"analyze-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"4"}},
|
|
{"type":"AudioAnalyzeFFT4096_IQ_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"FFT4096iq","inputs":2,"outputs":0,"category":"analyze-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"0"}},
|
|
|
|
{"type":"AudioAnalyzeFFT4096_IQem_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"FFT4096IQem","inputs":2,"outputs":0,"category":"analyze-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"0"}},
|
|
|
|
{"type":"AudioAnalyzeToneDetect_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"toneDetect","inputs":"1","output":"0","category":"analyze-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"0"}},
|
|
{"type":"AudioAnalyzeCTCSS_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"toneCTCSS","inputs":"1","output":"1","category":"analyze-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"1"}},
|
|
{"type":"AudioCalcEnvelope_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"calcEnvelope","inputs":"1","output":"0","category":"calc-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"1"}},
|
|
{"type":"AudioCalcGainWDRC_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"calcGainWDRC","inputs":"1","output":"0","category":"calc-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"1"}},
|
|
{"type":"AudioCalcLevel_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"calcLevel","inputs":"NaN","output":"0","category":"calc-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"NaN"}},
|
|
{"type":"AudioCalcGainWDRC_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"calcGainWDRC","inputs":"NaN","output":"0","category":"calc-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"1"}},
|
|
{"type":"AudioConvert_I16toF32","data":{"defaults":{"name":{"value":"new"}},"shortName":"convert_I16toF32","inputs":"0","output":"0","category":"convert-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"1"}},
|
|
{"type":"AudioConvert_F32toI16","data":{"defaults":{"name":{"value":"new"}},"shortName":"convert_F32toI16","inputs":"1","output":"0","category":"convert-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"0"}},
|
|
{"type":"AudioEffectCompWDRC_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"compWDRC","inputs":"1","output":"0","category":"effect-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"1"}},
|
|
{"type":"AudioEffectCompressor_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"compressor","inputs":"1","output":"0","category":"effect-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"1"}},
|
|
{"type":"AudioEffectDelay_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"delay","inputs":"1","output":"0","category":"effect-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"1"}},
|
|
{"type":"AudioEmpty_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"empty","inputs":"1","output":"0","category":"effect-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"1"}},
|
|
{"type":"AudioEffectCompressor2_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"compressor2","inputs":"1","output":"0","category":"effect-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"1"}},
|
|
{"type":"AudioEffectNoiseGate_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"noiseGate","inputs":"1","output":"0","category":"effect-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"1"}},
|
|
{"type":"AudioEffectFreqShiftFD_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"freqShift","inputs":"1","output":"0","category":"effect-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"1"}},
|
|
{"type":"AudioEffectGain_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"gain","inputs":"1","output":"0","category":"effect-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"1"}},
|
|
{"type":"AudioFilterFIRGeneral_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"filterFIRgeneral","inputs":"1","output":"0","category":"filter-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"1"}},
|
|
{"type":"AudioFilterEqualizer_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"filterEqualizer","inputs":"1","output":"0","category":"filter-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"1"}},
|
|
{"type":"AudioFilter90Deg_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"filter90deg","inputs":"2","output":"2","category":"filter-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"2"}},
|
|
{"type":"AudioFilterBiquad_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"biquad","inputs":"1","output":"0","category":"filter-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"1"}},
|
|
{"type":"AudioFilterFIR_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"fir","inputs":"1","output":"0","category":"filter-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"1"}},
|
|
|
|
|
|
{"type":"AudioFilterConvolution_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"convFilt","inputs":"1","output":"0","category":"filter-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"1"}},
|
|
|
|
|
|
{"type":"AudioLMSDenoiseNotch_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"LMS","inputs":"1","output":"0","category":"filter-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"1"}},
|
|
{"type":"AudioSpectralDenoise_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"Spectral","inputs":"1","output":"0","category":"filter-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"1"}},
|
|
{"type":"AudioFilterFreqWeighting_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"freqWeight","inputs":"NaN","output":"0","category":"filter-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"NaN"}},
|
|
{"type":"AudioFilterTimeWeighting_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"timeWeight","inputs":"1","output":"0","category":"filter-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"1"}},
|
|
{"type":"AudioMathAdd_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"mathAdd","inputs":"2","output":"0","category":"math-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"1"}},
|
|
{"type":"AudioMathMultiply_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"mathMultiply","inputs":"2","output":"0","category":"math-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"1"}},
|
|
{"type":"AudioMathOffset_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"mathOffset","inputs":"1","output":"0","category":"math-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"1"}},
|
|
{"type":"AudioMathScale_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"mathScale","inputs":"1","output":"0","category":"math-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"1"}},
|
|
{"type":"AudioMixer4_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"mixer4","inputs":"4","output":"0","category":"mixer-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"1"}},
|
|
{"type":"AudioMixer8_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"mixer8","inputs":"8","output":"0","category":"mixer-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"1"}},
|
|
{"type":"AudioSwitch4_OA_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"switch4","inputs":"1","output":"0","category":"mixer-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"4"}},
|
|
{"type":"AudioSwitch8_OA_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"switch8","inputs":"1","output":"0","category":"mixer-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"8"}},
|
|
{"type":"FFT_Overlapped_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"blockwiseFFT","inputs":"NaN","output":"0","category":"analyze-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"NaN"}},
|
|
{"type":"IFFT_Overlapped_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"blockwiseIFFT","inputs":"NaN","output":"0","category":"analyze-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"NaN"}},
|
|
{"type":"AudioInputI2S_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"audioInI2S","inputs":"0","output":"0","category":"input-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"2"}},
|
|
{"type":"AudioOutputI2S_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"audioOutI2S","inputs":"2","output":"0","category":"output-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"0"}},
|
|
{"type":"AudioInputUSB_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"audioInUSB","inputs":"0","output":"0","category":"input-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"2"}},
|
|
{"type":"AudioOutputUSB_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"audioOutUSB","inputs":"2","output":"0","category":"output-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"0"}},
|
|
{"type":"AudioPlayQueue_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"playQueue","inputs":"0","output":"0","category":"play-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"1"}},
|
|
{"type":"AudioRecordQueue_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"recordQueue","inputs":"1","output":"0","category":"record-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"0"}},
|
|
{"type":"AudioSynthNoisePink_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"noisePink","inputs":"0","output":"0","category":"synth-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"1"}},
|
|
{"type":"AudioSynthWaveformSine_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"sine","inputs":"0","output":"0","category":"synth-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"1"}},
|
|
{"type":"AudioSynthSineCosine_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"sine-cos","inputs":"0","output":"0","category":"synth-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"2"}},
|
|
{"type":"AudioSynthWaveform_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"waveform","inputs":"0","output":"0","category":"synth-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"1"}},
|
|
{"type":"AudioSynthNoiseWhite_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"noiseWhite","inputs":"0","output":"0","category":"synth-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"1"}},
|
|
{"type":"AudioSynthGaussian_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"GaussianWhiteNoise","inputs":"0","output":"0","category":"synth-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"1"}},
|
|
|
|
{"type":"AudioAlignLR_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"alignLR","inputs":"2","output":"0","category":"input-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"3"}},
|
|
|
|
{"type":"RadioFMDetector_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"FMDetector","inputs":"1","output":"0","category":"radio-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"2"}},
|
|
{"type":"radioModulatedGenerator_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"Modulator","inputs":"2","output":"0","category":"radio-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"2"}},
|
|
{"type":"radioNoiseBlanker_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"NoiseBlank","inputs":"2","output":"0","category":"radio-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"2"}},
|
|
{"type":"RadioIQMixer_F32","data":{"defaults":{"name":{"value":"new"}},"shortName":"I-QMixer","inputs":"2","output":"0","category":"radio-function","color":"#E6E0F8","icon":"arrow-in.png","outputs":"2"}},
|
|
{"type":"AudioControlSGTL5000","data":{"defaults":{"name":{"value":"new"}},"shortName":"sgtl5000","inputs":0,"outputs":0,"category":"control-function","color":"#E6E0F8","icon":"arrow-in.png"}},
|
|
{"type":"AudioControlAK4558","data":{"defaults":{"name":{"value":"new"}},"shortName":"ak4558","inputs":0,"outputs":0,"category":"control-function","color":"#E6E0F8","icon":"arrow-in.png"}},
|
|
{"type":"AudioControlCS4272","data":{"defaults":{"name":{"value":"new"}},"shortName":"cs4272","inputs":0,"outputs":0,"category":"control-function","color":"#E6E0F8","icon":"arrow-in.png"}},
|
|
{"type":"AudioControlWM8731","data":{"defaults":{"name":{"value":"new"}},"shortName":"wm8731","inputs":0,"outputs":0,"category":"control-function","color":"#E6E0F8","icon":"arrow-in.png"}},
|
|
{"type":"AudioControlWM8731master","data":{"defaults":{"name":{"value":"new"}},"shortName":"wm8731m","inputs":0,"outputs":0,"category":"control-function","color":"#E6E0F8","icon":"arrow-in.png"}},
|
|
{"type":"AudioControlCS42448","data":{"defaults":{"name":{"value":"new"}},"shortName":"cs42448","inputs":0,"outputs":0,"category":"control-function","color":"#E6E0F8","icon":"arrow-in.png"}}
|
|
|
|
]}
|
|
</script>
|
|
|
|
<script type="text/x-red" data-help-name="AudioCalcEnvelope_F32">
|
|
<p> AudioCalcEnvelope_F32</p>
|
|
<p> Created: Chip Audette, Feb 2017</p>
|
|
<p> Purpose: This module extracts the envelope of the audio signal.</p>
|
|
<p> Derived From: Core envelope extraction algorithm is from "smooth_env"</p>
|
|
<p> WDRC_circuit from CHAPRO from BTNRC: https://github.com/BTNRH/chapro</p>
|
|
<p> As of Feb 2017, CHAPRO license is listed as "Creative Commons?"</p>
|
|
<p> This processes a single stream fo audio data (ie, it is mono)</p>
|
|
<p>Used in support of other classes. Deprecated for use in an INO.
|
|
See Compressor and Compressor2 for complete, ready to use classes.</p>
|
|
<p> MIT License. use at your own risk.</p>
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioCalcEnvelope_F32 ">
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
<script type="text/x-red" data-help-name="AudioCalcGainWDRC_F32">
|
|
<p> AudioCalcGainWDRC_F32</p>
|
|
<p> Created: Chip Audette, Feb 2017</p>
|
|
<p> Purpose: This module calculates the gain needed for wide dynamic range compression.</p>
|
|
<p> Derived From: Core algorithm is from "WDRC_circuit"</p>
|
|
<p> WDRC_circuit from CHAPRO from BTNRC: https://github.com/BTNRH/chapro</p>
|
|
<p> As of Feb 2017, CHAPRO license is listed as "Creative Commons?"</p>
|
|
<p> This processes a single stream of audio data (ie, it is mono)</p>
|
|
<p>Used in support of other classes. Deprecated for use in an INO.
|
|
See Compressor and Compressor2 for complete, ready to use classes.</p>
|
|
<p> MIT License. use at your own risk.</p>
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioCalcGainWDRC_F32 ">
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
<script type="text/x-red" data-help-name="AudioCalcLevel_F32">
|
|
<p>Time weighting for sound level meter. Defaults to SLOW</p>
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioCalcLevel_F32 ">
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
<script type="text/x-red" data-help-name="AudioCalcGainWDRC_F32">
|
|
<p> AudioCalcGainWDRC_F32: Wide Dynamic Rnage Compressor</p>
|
|
<p> Created: Chip Audette (OpenAudio) Feb 2017</p>
|
|
<p> Derived From: WDRC_circuit from CHAPRO from BTNRC: https://github.com/BTNRH/chapro</p>
|
|
<p> As of Feb 2017, CHAPRO license is listed as "Creative Commons?"</p>
|
|
<p> MIT License. Use at your own risk.</p>
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioCalcGainWDRC_F32 ">
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
<script type="text/x-red" data-help-name="AudioEffectCompWDRC_F32">
|
|
<p> AudioCalcGainWDRC_F32: Wide Dynamic Rnage Compressor</p>
|
|
<p> Created: Chip Audette (OpenAudio) Feb 2017</p>
|
|
<p> Derived From: WDRC_circuit from CHAPRO from BTNRC: https://github.com/BTNRH/chapro</p>
|
|
<p> As of Feb 2017, CHAPRO license is listed as "Creative Commons?"</p>
|
|
<p> MIT License. Use at your own risk.</p>
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioEffectCompWDRC_F32 ">
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
<div>
|
|
<script type="text/x-red" data-help-name="AudioConvert_I16toF32">
|
|
<!-- ============ AudioConvert_I16toF32 ========= -->
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>Takes daudio data in conventional Teensy Audio integer format,
|
|
and converts it to floating point data suitable for this F32 Audio library.</p>
|
|
</div>
|
|
<h3>Boards Supported</h3>
|
|
<ul>
|
|
<li>Teensy 3.2
|
|
<li>Teensy 3.5
|
|
<li>Teensy 3.6
|
|
<li>Teensy 4.0
|
|
<li>Teensy 4.1
|
|
</ul>
|
|
<h3>Audio Connections</h3>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=top><th>Port</th><th>Purpose</th></tr>
|
|
<tr class=odd><td align=center>Out 0</td><td>Output F32 format</td></tr>
|
|
</table>
|
|
<h3>Functions</h3>
|
|
<p>There are no parameters to be modified and this class has no functions.
|
|
|
|
<h3>Examples</h3>
|
|
<p class=exam>File > Examples > OpenAudio_ArduinoLibrary > MyAudioEffect_Float
|
|
</p>
|
|
|
|
<h3>Notes</h3>
|
|
<p>This is the Teensy Audio I16 to OpenAudio_F32 data cconverter. The 16-bit signed
|
|
integer input can range from -32768 to 32767 in level The floating point (F32)
|
|
output scales this same range to -1.0 to 0.9999695 usually called "-1 to +1."
|
|
There are no limits of signal range on the floating point side, so -1 to +1
|
|
is chosen for convenience. The integer data has a dynamic range that is 8-bits
|
|
less than that of the floating point side, so no loss of precision
|
|
occurs with this conversion. The integer value 0 converts to the floating point
|
|
value, 0.0.</p>
|
|
|
|
<p>Only an output is shown for this converter because this Design Tool only follows
|
|
the flow of F32 floating point data. Work is in progress to allow
|
|
a mix of I16 and F32 in the same Design Tool.</p>
|
|
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioConvert_I16toF32">>
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
|
|
|
|
<div>
|
|
<script type="text/x-red" data-help-name="AudioConvert_F32toI16">
|
|
<!-- ============ AudioConvert_F32toI16 ========= -->
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>Takes audio data in F32 Audio library floating point format conventional
|
|
and converts it to Teensy Audio integer I16 format.</p>
|
|
</div>
|
|
<h3>Boards Supported</h3>
|
|
<ul>
|
|
<li>Teensy 3.2
|
|
<li>Teensy 3.5
|
|
<li>Teensy 3.6
|
|
<li>Teensy 4.0
|
|
<li>Teensy 4.1
|
|
</ul>
|
|
<h3>Audio Connections</h3>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=top><th>Port</th><th>Purpose</th></tr>
|
|
<tr class=odd><td align=center>In 0</td><td>Input F32 format</td></tr>
|
|
</table>
|
|
<h3>Functions</h3>
|
|
<p>There are no parameters to be modified and this class has no functions.
|
|
|
|
<h3>Examples</h3>
|
|
<p class=exam>File > Examples > OpenAudio_ArduinoLibrary > MyAudioEffect_Float
|
|
</p>
|
|
|
|
<h3>Notes</h3>
|
|
<p>This is the OpenAudio_F32 to Teensy Audio I16 data cconverter.
|
|
The floating point (F32) range must be limited to
|
|
a range of -1.0 to 0.9999695, usually called "-1 to +1," or
|
|
clipping will occur with this conversion. The 16-bit signed
|
|
integer output can only range from -32768 to 32767. Values outside
|
|
these limits will be clipped. The integer data will have a dynamic range
|
|
that is 8-bits less than that of the floating point side, and so a loss of precision
|
|
occurs with this conversion. The floating point vaue 0.0 converts
|
|
to an integer value of 0.
|
|
value, 0.0.</p>
|
|
|
|
|
|
<p>Only an input is shown for this converter because this Design Tool only follows
|
|
the flow of F32 floating point data. Work is in progress to allow
|
|
a mix of I16 and F32 in the same Design Tool.</p>
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioConvert_F32toI16">>
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
|
|
<div>
|
|
<script type="text/x-red" data-help-name="AudioFilterEqualizer_F32">
|
|
<!-- ============ AudioFilterEqualizer_F32 ========= -->
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>Uses flat-delay FIR filtering to generate a filter with
|
|
arbitrary amplitude response.</p>
|
|
</div>
|
|
<h3>Boards Supported</h3>
|
|
<ul>
|
|
<li>Teensy 3.6
|
|
<li>Teensy 4.0
|
|
<li>Teensy 4.1
|
|
</ul>
|
|
<h3>Audio Connections</h3>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=top><th>Port</th><th>Purpose</th></tr>
|
|
<tr class=odd><td align=center>In 0</td><td>Input Signal</td></tr>
|
|
<tr class=odd><td align=center>Out 0</td><td>Output Equalized Signal</td></tr>
|
|
</table>
|
|
<h3>Functions</h3>
|
|
|
|
<p class=func><span class=keyword>equalizerNew</span>(<strong>uint16_t</strong> nBands, <strong>float</strong> *feq, <strong>float</strong> *adb, <strong>uint16_t</strong> nFIR, <strong>float</strong> *cf32f, <strong>float</strong> kdb);</p>
|
|
<p class=desc>Does the design of the equalizer and enables it for audio updates.
|
|
nBands is the number of frequecy bands,
|
|
feq points to an array of band tops in Hz,
|
|
adb points to an array of dB levels for the bands,
|
|
nFIR is the number of FIR coefficients being used and
|
|
kdb is the Kaiser window parameter that sets the sidelobe response.
|
|
</p>
|
|
|
|
<p class=func><span class=keyword>getResponse</span>(<strong>uint16_t</strong> nFreq, <strong>float</strong> *rdb);</p>
|
|
<p class=desc>Calculates the response of the equalizer in dB at nFreq equally spaced
|
|
frequencies. rdb is a pointer to an array of nFreq floats where the response can be put.
|
|
</p>
|
|
|
|
<h3>Examples</h3>
|
|
<p class=exam>File > Examples > OpenAudio_ArduinoLibrary > TestEqualizer1
|
|
</p>
|
|
|
|
<p class=exam>File > Examples > OpenAudio_ArduinoLibrary > TestEqualizer1Audio
|
|
</p>
|
|
|
|
<h3>Notes</h3>
|
|
<p>This equalizer is specified by an array of 'nBands' frequency bands
|
|
each of of arbitrary frequency span. The first band always starts at
|
|
0.0 Hz, and that value is not entered. Each band is specified by the upper
|
|
frequency limit to the band.
|
|
The last band always ends at half of the sample frequency, which for 44117 Hz
|
|
sample frequency would be 22058.5. Each band is specified by its upper
|
|
frequency in an .INO supplied array feq[]. The dB level of that band is
|
|
specified by a value, in dB, arranged in an .INO supplied array
|
|
aeq[]. Thus a trivial bass/treble control might look like:</p>
|
|
<pre class="desc">
|
|
nBands = 3;
|
|
feq[] = {300.0, 1500.0, 22058.5};
|
|
float32_t bass = -2.5; // in dB, relative to anything
|
|
float32_t treble = 6.0;
|
|
aeq[] = {bass, 0.0, treble};
|
|
</pre>
|
|
<p>Note that there is much, much more information in the
|
|
AudioFilterEqualizer_F32.h file at the OpenAudio_Arduino directory.
|
|
This includes more examples.</p>
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioFilterEqualizer_F32">>
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
|
|
|
|
<div>
|
|
<script type="text/x-red" data-help-name="AudioFilterFIRGeneral_F32">
|
|
<!-- ============ AudioFilterFIRGeneral_F32 ========= -->
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>Uses flat-delay FIR filtering to generate Low Pass, High Pass,
|
|
Band Pas and Band Reject Filters, do the filtering and compute the
|
|
response.</p>
|
|
</div>
|
|
<h3>Boards Supported</h3>
|
|
<ul>
|
|
<li>Teensy 3.5
|
|
<li>Teensy 3.6
|
|
<li>Teensy 4.0
|
|
<li>Teensy 4.1
|
|
</ul>
|
|
<h3>Audio Connections</h3>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=top><th>Port</th><th>Purpose</th></tr>
|
|
<tr class=odd><td align=center>In 0</td><td>Input Signal</td></tr>
|
|
<tr class=odd><td align=center>Out 0</td><td>Output Filtered Signal</td></tr>
|
|
</table>
|
|
<h3>Functions</h3>
|
|
|
|
<p class=func><span class=keyword>FIRGeneralNew</span>(<strong>float</strong> *adb, <strong>uint16_t</strong> nFIR, <strong>float</strong> *cf32f, <strong>float</strong> kdb, <strong>float</strong> *pStateArray);</p>
|
|
<p class=desc>Does the design of the filter and enables it for audio updates.
|
|
adb points to an array of dB levels for the bands,
|
|
nFIR is the number of FIR coefficients being used,
|
|
cf32f is a pointer to an INO supplied array where the coefficients will be stored,
|
|
nBands is the number of frequecy bands,
|
|
feq points to an array of band tops in Hz,
|
|
kdb is the Kaiser window parameter that sets the sidelobe response, and
|
|
pStateArray is a pointer to an INO supplied 2*nFIR+128 array of floats for working space.
|
|
</p>
|
|
|
|
<p class=func><span class=keyword>LoadCoeffs</span>(<strong>uint16_t</strong> nFIR, <strong>float</strong> *cf32, <strong>float</strong> pStateArray);</p>
|
|
<p class=desc>Loads new filter coefficients that are INO supplied.
|
|
nFIR is the number of FIR coefficients being used,
|
|
cf32 is a pointer to an array of FIR coefficients (name of array)
|
|
pStateArray is a pointer to a 2*nFIR+128 array of floats for working space (name of array)
|
|
</p>
|
|
|
|
<p class=func><span class=keyword>getResponse</span>(<strong>uint16_t</strong> nFreq, <strong>float</strong> *rdb);</p>
|
|
<p class=desc>Calculates the response of the equalizer in dB at nFreq equally spaced
|
|
frequencies. rdb is a pointer to an array of nFreq floats where the response can be put.</p>
|
|
|
|
<h3>Examples</h3>
|
|
<p class=exam>File > Examples > OpenAudio_ArduinoLibrary > TestFIRGeneralLarge4
|
|
</p>
|
|
|
|
<p class=exam>File > Examples > OpenAudio_ArduinoLibrary > TestFIRGeneralLarge5
|
|
</p>
|
|
<h3>Notes</h3>
|
|
|
|
<p>For those new to C, when a pointer to an array is needed in a function call, just
|
|
supply the name of the array. See the examples to see how this works.</p>
|
|
|
|
<p>The frequency response of the filter is specfied by a series of ideal dB levels.
|
|
for instance, 0.0 dB for a pass band and -150.0 dB for a stop band.
|
|
IF nFIR is even, the response specification needs nFIR/2 points and if nFIR is odd,
|
|
it needs (nFIR+1)/2 points. The "to do" list has macros to specify the common
|
|
responses. Meanwhile, see the example INOs for methods of doing this. </p>
|
|
|
|
<p>The INO must supply memory that is used by the ARM FIR routine called workspace.
|
|
This is an array of float variables that is 2*nFIR+128 in size. The call to design and
|
|
use a filter need a pointer to this array, which for C is the name of the array.</p>
|
|
|
|
<p>The header file, AudioFilterFIRGeneral_F32.h in the OpenAudio_ArduinoLibrary directory
|
|
includes many notes, timing and examples. Be sure to look at those when getting
|
|
acquainted with this class.</p>
|
|
|
|
<p>FIR filters can be (and are here) implemented to have symmetrical coefficients. This
|
|
results in constant delay at all frequencies (linear phase). For some applications this can
|
|
be an important feature. Sometimes it is suggested that the FIR should not be
|
|
used because of the latency it creates. Note that if constant delay is needed, the FIR
|
|
implementation does this with minimum latency.</p>
|
|
|
|
<p>AudioFilterFIR_F32 in this OpenAudio_ArduinoLibrary handles 32-bit floating point
|
|
data and a maximum of 200 taps. This class requires the INO to provide the working
|
|
space and thereby puts no limit on the number of FIR taps (coefficients) being used.
|
|
The processor does run out of time, and that limits Teensy 3.6 to about 6000 taps
|
|
and Teensy 4.x to about 6000. As a starting spot for huge FIR filters, one might use
|
|
1/2 or 1/3 of those numbers.</p>
|
|
|
|
<p>It is practical to switch filter coefficient arrays on-the-fly.
|
|
See the LoadCoeffs() function. This class is initialized to a 4 coefficient
|
|
pass-through filter.</p>
|
|
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioFilterFIRGeneral_F32">>
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
<div>
|
|
<script type="text/x-red" data-help-name="AudioFilterConvolution_F32">
|
|
<!-- ============ AudioFilterConvolution_F32 ========= -->
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p> Convolutional filtering. Faster than a FIR if you want a 512 tap filter.</p>
|
|
</div>
|
|
<h3>Boards Supported</h3>
|
|
<ul>
|
|
<li>Teensy 3.5
|
|
<li>Teensy 3.6
|
|
<li>Teensy 4.0
|
|
<li>Teensy 4.1
|
|
</ul>
|
|
<h3>Audio Connections</h3>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=top><th>Port</th><th>Purpose</th></tr>
|
|
<tr class=odd><td align=center>In 0</td><td>Input Signal</td></tr>
|
|
<tr class=odd><td align=center>Out 0</td><td>Output Filtered Signal</td></tr>
|
|
</table>
|
|
<h3>Functions</h3>
|
|
|
|
<p class=func><span class=keyword>initFilter</span>(<strong>void</strong>);</p>
|
|
<p class=desc> Without parameters, the initFilter function does not design a filter, but rather
|
|
just uses whatever FIR coefficients are in place. These may have been
|
|
loaded by way of the getCoeffPtr() function or passThrough(1) may be in place. </p>
|
|
|
|
<p class=func><span class=keyword>initFilter</span>(<strong>float32_t</strong> fc,
|
|
<strong>float32_t</strong> Astop, <strong>int</strong> type, <strong>float32_t</strong> dfc);</p>
|
|
<p class=desc> Designs filters. fc is the edge frequency in Hz for LPF and HPF
|
|
and center frequency for BPF and BRF, Astop is the stopband attenuation in dB.
|
|
The parameter dfc is the filter bandwidth (only for bandpass and band reject).
|
|
The parameter type sets the filter per the following defines:
|
|
<pre class="desc">
|
|
LOWPASS Low pass with fc cutoff frequency
|
|
HIGHPASS High pass with fc cutoff frequency.
|
|
BANDPASS Band pass with fc center frequency and dfc pass band width.
|
|
BANDREJECT Band reject with fc center frequency and dfc reject band width.
|
|
HILBERT Hilbert transform. Not Currently Available
|
|
</pre>
|
|
</p>
|
|
|
|
<p class=func><span class=keyword>passThrough</span>(<strong>int</strong>stat)</p>
|
|
<p class=desc>passThrough(int stat) allows data for this filter object to be passed through
|
|
unchanged with stat=1. The default is stat=0.
|
|
</p>
|
|
|
|
<p class=func><span class=keyword>getCoeffPtr</span>(<strong>void</strong>);</p>
|
|
<p class=desc>Returns a pointer to the coefficient array. To use this, compute
|
|
the coefficients of a 512 tap FIR filter with the desired response. Then
|
|
load the 512 float32_t buffer with the coefficients.
|
|
</p>
|
|
|
|
<h3>Examples</h3>
|
|
<p class=exam>File > Examples > OpenAudio_ArduinoLibrary > TestConvolutionFilter_F32
|
|
</p>
|
|
|
|
|
|
<h3>Notes</h3>
|
|
|
|
<p>See the file AudioFilterConvolution_F32.h, as well as the example file, just above, for many notes,
|
|
hints and sample code.</p>.
|
|
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioFilterConvolution_F32">>
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
<div>
|
|
<script type="text/x-red" data-help-name="AudioLMSDenoiseNotch_F32">
|
|
<!-- ============ AudioLMSDenoiseNotch_F32 ========= -->
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>Provides LMS denoise or auto-notch functions.</p>
|
|
</div>
|
|
<h3>Boards Supported</h3>
|
|
<ul>
|
|
<li>Teensy 3.6
|
|
<li>Teensy 4.0
|
|
<li>Teensy 4.1
|
|
</ul>
|
|
<h3>Audio Connections</h3>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=top><th>Port</th><th>Purpose</th></tr>
|
|
<tr class=odd><td align=center>In 0</td><td>Input Signal</td></tr>
|
|
<tr class=odd><td align=center>Out 0</td><td>Output Filtered Signal</td></tr>
|
|
</table>
|
|
<h3>Functions</h3>
|
|
<p class=func><span class=keyword>initializeLMS</span>(<strong>uint16_t</strong> what, <strong>uint16_t</strong> lengthDataF, <strong>uint16_t</strong> lengthDataD);</p>
|
|
<p class=desc>The parameter what must be either DENOISE or NOTCH. The lengthDataF buffer should
|
|
be a power of 2 between 2 and 128. It will be shifted down to a power of 2, if not. The delay
|
|
buffer size, lengthDataD can be any value between 1 and 16.
|
|
</p>
|
|
|
|
<p class=func><span class=keyword>setParameters</span>(<strong>float32_t</strong> beta, <strong>float32_t</strong> decay);</p>
|
|
<p class=desc>Sets the gain, beta, that varies the amount of denoise and notching. Adjust
|
|
for the desired sound.
|
|
</p>
|
|
|
|
<p class=func><span class=keyword>enable</span>(<strong>bool</strong> setEnable);</p>
|
|
<p class=desc>The parameter setEnable should be <strong>true</strong> or <strong>false.</strong>
|
|
If not enabled, the audio is passed through unchanged.
|
|
</p>
|
|
|
|
<h3>Examples</h3>
|
|
<p class=exam>File > Examples > OpenAudio_ArduinoLibrary > LMS1.ino
|
|
</p>
|
|
|
|
<h3>Notes</h3>
|
|
<p>This block can be changed at any time between denoise and auto-notching. Thus, usually
|
|
only one of these is used in a radio.</p>
|
|
<p>More notes are included in the file AudioLMSDenoiseNotch_F32.h</p>
|
|
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioLMSDenoiseNotch_F32">>
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
<div>
|
|
<script type="text/x-red" data-help-name="AudioSpectralDenoise_F32">
|
|
<!-- ============ AudioSpectralDenoise_F32 ========= -->
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>Spectral Noise Reduction</p>
|
|
<p>Remove random noise, such as background noise or machine generated noise, and try
|
|
to leave voice intact.</p>
|
|
<p>Spectral NR does not try to remove constant tones or clicks and pops.</p>
|
|
<p>Spectral NR is generally not useful for cleaning up 'pure tone' sources, such as morse code.</p>
|
|
<h3>Audio Connections</h3>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=top><th>Port</th><th>Purpose</th></tr>
|
|
<tr class=odd><td align=center>In 0</td><td>Signal to be filtered</td></tr>
|
|
<tr class=odd><td align=center>Out 0</td><td>Filtered Signal Output</td></tr>
|
|
</table>
|
|
<h3>Functions</h3>
|
|
<p class=func><span class=keyword>setup</span>(<strong>AudioSettings_F32 &</strong>settings, <strong>int</strong> n_fft);</p>
|
|
<p class=desc>Setup the filter. Must be called if parameters not passed in to the
|
|
constructor. Can be re-called if other parameters have been changed to have
|
|
the changes take effect.
|
|
The maximum FFT size is dictated by the underlying FFT_OA code.
|
|
</p>
|
|
<p class=func><span class=keyword>enable</span>(<strong>bool</strong> start);</p>
|
|
<p class=desc>Turn on or offthe filter. The filter is enabled by default upon
|
|
creation.
|
|
</p>
|
|
<p class=func><span class=keyword>enabled</span>();</p>
|
|
<p class=desc>Return the enabled state of the filter.</p>
|
|
<p class=func><span class=keyword>getAsnr</span>();</p>
|
|
<p class=desc>Return the current ASNR value.</p>
|
|
<p class=func><span class=keyword>setAsnr</span>(<strong>float32_t</strong> value);</p>
|
|
<p class=desc>Set the ASNR value. Active SNR.</p>
|
|
<p class=desc>Call <span class=keyword>setup()</span> for the change to take effect.</p>
|
|
<p class=func><span class=keyword>getVADHighFreq</span>();</p>
|
|
<p class=desc>Get the VAD band high frequency cutoff value.</p>
|
|
<p class=func><span class=keyword>setVADHighFreq</span>(<strong>float32_t</strong> value);</p>
|
|
<p class=desc>Set the VAD High Frequency cutoff value. Sets the high
|
|
frequency value of the Voice Activity Detector (VAD) window.
|
|
</p>
|
|
<p class=desc>Call <span class=keyword>setup()</span> for the change to take effect.</p>
|
|
<p class=func><span class=keyword>getVADLowFreq</span>();</p>
|
|
<p class=desc>Get the VAD band low frequency value.</p>
|
|
<p class=func><span class=keyword>setVADLowFreq</span>(<strong>float32_t</strong> value);</p>
|
|
<p class=desc>Set the VAD band low frequency value.</p>
|
|
<p class=desc>Call <span class=keyword>setup()</span> for the change to take effect.</p>
|
|
<p class=func><span class=keyword>getNRAlpha</span>();</p>
|
|
<p class=desc>get the NRalpha value. NR alpha is the time smoothing constant. It's
|
|
range is clipped between 0.9 and 0.9999 in the code. Settings >0.95
|
|
generally recommended, or you can get strong reverb and 'watery' effects.
|
|
</p>
|
|
<p class=func><span class=keyword>setNRAlpha</span>(<strong>float32_t</strong> value);</p>
|
|
<p class=desc>Set the NRalpha value.</p>
|
|
<p class=desc>The change takes effect immediately.</p>
|
|
<p class=func><span class=keyword>getSNRPrioMin</span>();</p>
|
|
<p class=desc>Get the SNR prior value.</p>
|
|
<p class=func><span class=keyword>setSNRPrioMin</span>(<strong>float32_t</strong> value);</p>
|
|
<p class=desc>Set the SNR prior value.</p>
|
|
<p class=desc>The change takes effect immediately.</p>
|
|
<p class=func><span class=keyword>getNRWidth</span>();</p>
|
|
<p class=desc>Get the NR width. Used for the measuring 'span' in the musical note
|
|
reduction code.
|
|
</p>
|
|
<p class=func><span class=keyword>setNRWidth</span>(<strong>int16_t</strong> value);</p>
|
|
<p class=desc>Set the NR width.</p>
|
|
<p class=desc>The change takes effect immediately.</p>
|
|
<p class=func><span class=keyword>getPowerThreshold</span>();</p>
|
|
<p class=desc>Get the power threshold. Used to limit the effects of the musical
|
|
note reduction code.
|
|
</p>
|
|
<p class=func><span class=keyword>setPowerThreshold</span>(<strong>float32_t</strong> value);</p>
|
|
<p class=desc>Set the power threshold.</p>
|
|
<p class=desc>The change takes effect immediately.</p>
|
|
<p class=func><span class=keyword>getTaxFactor</span>();</p>
|
|
<p class=desc>Get the noise output smoothing factor.</p>
|
|
<p class=func><span class=keyword>setTaxFactor</span>(<strong>float32_t</strong> value);</p>
|
|
<p class=desc>Set the noise output smoothing factor. Typical values are around 0.8.</p>
|
|
<p class=desc>Call <span class=keyword>setup()</span> for the change to take effect.</p>
|
|
<p class=func><span class=keyword>getTapFactor</span>();</p>
|
|
<p class=desc>Get the speech probability smoothing factor.</p>
|
|
<p class=func><span class=keyword>setTapFactor</span>(<strong>float32_t</strong> value);</p>
|
|
<p class=desc>Set the speech probability smoothing factor. Typical values are around 0.9.</p>
|
|
<p class=desc>Call <span class=keyword>setup()</span> for the change to take effect.</p>
|
|
<h3>Examples</h3>
|
|
<p class=exam>File > Examples > OpenAudio_ArduinoLibrary > SpectralDenoise.ino
|
|
</p>
|
|
<h3>Notes</h3>
|
|
<p> The defaults work OK, but it is worth experimenting with the adjustable
|
|
parameters for your specific situation. Try adjusting NRAlpha, NRWidth and Asnr.
|
|
</p>
|
|
<p> The default FFT size is 256. You may wish to make this larger, which will allow
|
|
a finer granularity of 'buckets' to potentially improve the noise reduction, but,
|
|
will cost both processor time and memory overheads.
|
|
</p>
|
|
<p> The maximum FFT size is set by the underlying <strong>FFT_OA</strong> code, which at
|
|
time of writing limits the maximum to 2048.
|
|
</p>
|
|
<p> The code is written to account for the <strong>sample_rate_Hz</strong> passed in with
|
|
the audio settings, <strong>but</strong>, this has not been tested. If you do test,
|
|
either with a higher or lower sample rate, please report back and consider fixing
|
|
this documentation.
|
|
</p>
|
|
<h3>References</h3>
|
|
<p>The best reference on how the Spectral code was designed and works can be found on the
|
|
<a href="https://github.com/df8oe/UHSDR/wiki/Noise-reduction"> UHSDR wiki </a>
|
|
</p>
|
|
</script>
|
|
</div>
|
|
|
|
<div>
|
|
<script type="text/x-red" data-help-name="AudioFilter90Deg_F32">
|
|
<!-- ============ AudioFilter90Deg_F32 ========= -->
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>Creates two uncoupled paths that almost have the same amplitude gain
|
|
but differ in phase by exactly 90 degrees. </p>
|
|
</div>
|
|
<h3>Boards Supported</h3>
|
|
<ul>
|
|
<li>Teensy 3.6
|
|
<li>Teensy 4.0
|
|
<li>Teensy 4.1
|
|
</ul>
|
|
<h3>Audio Connections</h3>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=top><th>Port</th><th>Purpose</th></tr>
|
|
<tr class=odd><td align=center>In 0</td><td>Input Signal Hilbert Filtered</td></tr>
|
|
<tr class=odd><td align=center>In 1</td><td>Input Signal Delayed</td></tr>
|
|
<tr class=odd><td align=center>Out 0</td><td>Output Signal Hilbert Filtered</td></tr>
|
|
<tr class=odd><td align=center>Out 1</td><td>Output Signal Delayed</td></tr>
|
|
</table>
|
|
<h3>Functions</h3>
|
|
|
|
<p class=func><span class=keyword>begin</span>(<strong>float</strong> *pCoeff, <strong>int</strong> nCoeff);</p>
|
|
<p class=desc>Initializes this block, with pCoeff being a pointer to array of F32 Hilbert Transform coefficients,
|
|
and nCoeff being the number of Hilbert transform coefficients (odd).
|
|
</p>
|
|
|
|
<h3>Examples</h3>
|
|
<p class=exam>File > Examples > OpenAudio_ArduinoLibrary > ReceiverPart1
|
|
</p>
|
|
|
|
<p class=exam>File > Examples > OpenAudio_ArduinoLibrary > ReceiverPart2
|
|
</p>
|
|
|
|
<h3>Notes</h3>
|
|
<p>This consists of two uncoupled paths that almost have the same amplitude gain
|
|
but differ in phase by exactly 90 degrees.
|
|
The number of coefficients is an odd number for the FIR Hilbert transform
|
|
as that produces an easily achievable integer sample period delay.</p>
|
|
|
|
<p>No default Hilbert Transform is provided, as it is highly application dependent.
|
|
The number of coefficients is an odd number with a maximum of 250. The Iowa
|
|
Hills program can design such a Hilbert Transform filter.</p>
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioFilter90Deg_F32">>
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
|
|
<!-- ============ AudioEffectCompressor_F32 ========= -->
|
|
<script type="text/x-red" data-help-name="AudioEffectCompressor_F32">
|
|
<p> AudioEffectCompressor</p>
|
|
<p> Created: Chip Audette, Dec 2016 - Jan 2017</p>
|
|
<p> Purpose; Apply dynamic range compression to the audio stream.</p>
|
|
<p> Assumes floating-point data.</p>
|
|
<p> This processes a single stream fo audio data (ie, it is mono)</p>
|
|
<p> MIT License. use at your own risk.</p>
|
|
<p> Note: This help documentation is incomplete. See AudioEffectCompressor2_F32
|
|
for a similar block, with documentation. Compressor2 includes up to 5 segments and
|
|
look ahead delay, as well.</p>
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioEffectCompressor_F32 ">
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
<!-- ============ AudioEffectDelay_F32 ========= -->
|
|
<script type="text/x-red" data-help-name="AudioEffectDelay_F32">
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>Delay a signal. Up to 8 separate delay taps can be used.</p>
|
|
<p align=center><img src="img/delay.png"><br><small>1 kHz burst, delayed 5.2 ms.</small></p>
|
|
</div>
|
|
<h3>Audio Connections</h3>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=top><th>Port</th><th>Purpose</th></tr>
|
|
<tr class=odd><td align=center>In 0</td><td>Signal Input</td></tr>
|
|
<tr class=odd><td align=center>Out 0</td><td>Delay Tap #1</td></tr>
|
|
<tr class=odd><td align=center>Out 1</td><td>Delay Tap #2</td></tr>
|
|
<tr class=odd><td align=center>Out 2</td><td>Delay Tap #3</td></tr>
|
|
<tr class=odd><td align=center>Out 3</td><td>Delay Tap #4</td></tr>
|
|
<tr class=odd><td align=center>Out 4</td><td>Delay Tap #5</td></tr>
|
|
<tr class=odd><td align=center>Out 5</td><td>Delay Tap #6</td></tr>
|
|
<tr class=odd><td align=center>Out 6</td><td>Delay Tap #7</td></tr>
|
|
<tr class=odd><td align=center>Out 7</td><td>Delay Tap #8</td></tr>
|
|
</table>
|
|
<h3>Functions</h3>
|
|
<p class=func><span class=keyword>delay</span>(channel, milliseconds);</p>
|
|
<p class=desc>Set output channel (0 to 7) to delay the signals by
|
|
milliseconds. The maximum delay is approx 425 ms. The actual delay
|
|
is rounded to the nearest sample. Each channel can be configured for
|
|
any delay. There is no requirement to configure the "taps" in increasing
|
|
delay order.
|
|
</p>
|
|
<p class=func><span class=keyword>disable</span>(channel);</p>
|
|
<p class=desc>Disable a channel. The output of this channel becomes
|
|
silent. If this channel is the longest delay, memory usage is
|
|
automatically reduced to accomodate only the remaining channels used.
|
|
</p>
|
|
<h3>Examples</h3>
|
|
<p class=exam>File > Examples > Audio > Effects > Delay
|
|
</p>
|
|
<h3>Notes</h3>
|
|
<p>Memory for the delayed signal is take from the memory pool allocated by
|
|
<a href="http://www.pjrc.com/teensy/td_libs_AudioConnection.html" target="_blank">AudioMemory()</a>.
|
|
Each block allows about 3 milliseconds of delay, so AudioMemory
|
|
should be increased to allow for the longest delay tap.
|
|
</p>
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioEffectDelay_F32">
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
|
|
<div>
|
|
<script type="text/x-red" data-help-name="AudioEffectFreqShiftFD_F32">
|
|
<!-- ============ AudioEffectFreqShiftFD_F32 ========= -->
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>This shifts frequencies of entire audio spectrums by use of FFT and IFFT
|
|
with bin shifting. </p>
|
|
</div>
|
|
<h3>Boards Supported</h3>
|
|
<ul>
|
|
<li>Teensy 3.6
|
|
<li>Teensy 4.0
|
|
<li>Teensy 4.1
|
|
</ul>
|
|
<h3>Audio Connections</h3>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=top><th>Port</th><th>Purpose</th></tr>
|
|
<tr class=odd><td align=center>In 0</td><td>Input Signal</td></tr>
|
|
<tr class=odd><td align=center>Out 0</td><td>Shifted Output Signal</td></tr>
|
|
</table>
|
|
<h3>Functions</h3>
|
|
|
|
<p class=func><span class=keyword>setup</span>(<strong>AudioSettings_F32 </strong> &settings, <strong>int</strong> N_FFT);</p>
|
|
<p class=desc>Initializes the needed FFTs. Returns the size of FFT achieved
|
|
or a negative number if unsuccessful. See example below for determining N_FFT.
|
|
</p>
|
|
|
|
<p class=func><span class=keyword>setShift_bins</span>(<strong>int</strong> nBins);</p>
|
|
<p class=desc>Sets nBins, the number of FFT bins of frequency shift.
|
|
</p>
|
|
|
|
<h3>Examples</h3>
|
|
<p class=exam>File > Examples > OpenAudio_ArduinoLibrary > FrequencyShifter_FD_OA
|
|
</p>
|
|
|
|
<h3>Notes</h3>
|
|
<p>Created: Chip Audette (OpenAudio) Aug 2019</p>
|
|
|
|
<p>This processing is performed in the frequency domain.
|
|
Frequencies can only be shifted by an integer number of bins,
|
|
so small frequency shifts are not possible. For example, for
|
|
a sample rate of 44.1kHz, and when using N=256, one can only
|
|
shift frequencies in multiples of 44.1/256 = 172.3 Hz.</p>
|
|
|
|
<p>This processing is performed in the frequency domain where
|
|
we take the FFT, shift the bins upward or downward, take
|
|
the IFFT, and listen to the results. In effect, this is
|
|
single sideband modulation, which will sound very unnatural
|
|
(like robot voices!). Maybe you'll like it, or maybe not.
|
|
Probably not, unless you like weird. ;) </p>
|
|
|
|
<p>You can shift frequencies upward or downward with this algorithm.</p>
|
|
<p>This class supports changes in block size by AudioSettings_F32.</p>
|
|
<p> Also see an I-Q mixing approach with fine tuning in the example: </p>
|
|
<p class=exam>File > Examples > OpenAudio_ArduinoLibrary > FineFreqShift_OA
|
|
</p>
|
|
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioEffectFreqShiftFD_F32">>
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
|
|
|
|
<div>
|
|
<script type="text/x-red" data-help-name="AudioSwitch4_OA_F32">
|
|
<!-- ============ AudioSwitch4_OA_F32 ========= -->
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>Switch one input to 4 outputs, which only triggers 1 of the 4
|
|
audio processing paths (thus saving computation on paths that you aren't
|
|
using). Use with mixer-4 for up to 4 paths of alternate processing.</p>
|
|
</div>
|
|
<h3>Boards Supported</h3>
|
|
<ul>
|
|
<li>Teensy 3.2
|
|
<li>Teensy 3.5
|
|
<li>Teensy 3.6
|
|
<li>Teensy 4.0
|
|
<li>Teensy 4.1
|
|
</ul>
|
|
<h3>Audio Connections</h3>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=top><th>Port</th><th>Purpose</th></tr>
|
|
<tr class=odd><td align=center>In 0</td><td>Input Signal 0</td></tr>
|
|
<tr class=odd><td align=center>Out 0</td><td>Switch Output Signal 0</td></tr>
|
|
<tr class=odd><td align=center>Out 1</td><td>Switch Output Signal 1</td></tr>
|
|
<tr class=odd><td align=center>Out 2</td><td>Switch Output Signal 2</td></tr>
|
|
<tr class=odd><td align=center>Out 3</td><td>Switch Output Signal 3</td></tr>
|
|
</table>
|
|
|
|
<h3>Functions</h3>
|
|
<p class=func><span class=keyword>setChannel</span>(<strong>int</strong> channel);</p>
|
|
<p class=desc>Selects the output path that will be active, 0 to 7. Returns the selected path, if valid.
|
|
</p>
|
|
|
|
<h3>Examples</h3>
|
|
<p class=exam>File > Examples > OpenAudio_ArduinoLibrary > Switches_float
|
|
</p>
|
|
|
|
<h3>Notes</h3>
|
|
<p>If you have two audio paths, like a SSB receiver path and an AM receiver path, you can
|
|
shut off the data going to the unused path with this AudioSwitch4_F32 class.
|
|
The AudioStream will not supply the packet to the next
|
|
object and that, in turn, will cause it to not request a transmission to the next
|
|
thing down the chain. The AudioStream routines do not create errors, it turns out,
|
|
and you save all the processing time and memory usage for the unused paths.</p>
|
|
|
|
<p>There is an 8 output version in the palette, if you need more outputs.</p>
|
|
|
|
<p>Created: Chip Audette, OpenAudio, April 2019</p>
|
|
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioSwitch4_OA_F32">>
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
<div>
|
|
<script type="text/x-red" data-help-name="AudioSwitch8_OA_F32">
|
|
<!-- ============ AudioSwitch8_OA_F32 ========= -->
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>Switch one input to 8 outputs, which only triggers 1 of the 8
|
|
audio processing paths (thus saving computation on paths that you aren't
|
|
using). Use with mixer-8 for up to 8 paths of alternate processing.</p>
|
|
</div>
|
|
<h3>Boards Supported</h3>
|
|
<ul>
|
|
<li>Teensy 3.2
|
|
<li>Teensy 3.5
|
|
<li>Teensy 3.6
|
|
<li>Teensy 4.0
|
|
<li>Teensy 4.1
|
|
</ul>
|
|
<h3>Audio Connections</h3>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=top><th>Port</th><th>Purpose</th></tr>
|
|
<tr class=odd><td align=center>In 0</td><td>Input Signal 0</td></tr>
|
|
<tr class=odd><td align=center>Out 0</td><td>Switch Output Signal 0</td></tr>
|
|
<tr class=odd><td align=center>Out 1</td><td>Switch Output Signal 1</td></tr>
|
|
<tr class=odd><td align=center>Out 2</td><td>Switch Output Signal 2</td></tr>
|
|
<tr class=odd><td align=center>Out 3</td><td>Switch Output Signal 3</td></tr>
|
|
<tr class=odd><td align=center>Out 4</td><td>Switch Output Signal 4</td></tr>
|
|
<tr class=odd><td align=center>Out 5</td><td>Switch Output Signal 5</td></tr>
|
|
<tr class=odd><td align=center>Out 6</td><td>Switch Output Signal 6</td></tr>
|
|
<tr class=odd><td align=center>Out 7</td><td>Switch Output Signal 7</td></tr>
|
|
</table>
|
|
|
|
<h3>Functions</h3>
|
|
<p class=func><span class=keyword>setChannel</span>(<strong>int</strong> channel);</p>
|
|
<p class=desc>Selects the output path that will be active, 0 to 7. Returns the selected path, if valid.
|
|
</p>
|
|
|
|
<h3>Examples</h3>
|
|
<p class=exam>File > Examples > OpenAudio_ArduinoLibrary > Switches_float
|
|
</p>
|
|
|
|
<h3>Notes</h3>
|
|
<p>If you have multiple audio paths, like 5 different demodulators, you can
|
|
shut off the data going to the unused paths with this AudioSwitch8_F32 class.
|
|
The AudioStream will not supply the packet to the next
|
|
object and that, in turn, will cause it to not request a transmission to the next
|
|
thing down the chain. The AudioStream routines do not create errors, it turns out,
|
|
and you save all the processing time and memory usage for the unused paths.</p>
|
|
|
|
<p>There is an 4 output version in the palette, if you need less outputs.</p>
|
|
|
|
<p>Created: Chip Audette, OpenAudio, April 2019</p>
|
|
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioSwitch8_OA_F32">>
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
|
|
|
|
<div>
|
|
<script type="text/x-red" data-help-name="AudioAnalyzePhase_F32">
|
|
<!-- <h2> ============ AudioAnalyzePhase_F32 AudioAnalyzePhase_F32 =========</h2> -->
|
|
<h3>AudioAnalyzePhase_F32</h3>
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>This block can be used to measure phase between two sinusoid inputs.
|
|
The default output IIR filter, 4-cascaded stages of BiQuad,
|
|
is suitable for this with a cut-off
|
|
frequency of 100 Hz. As an alternative, a linear phase
|
|
FIR filter can be set up with the begin function. The built in FIR
|
|
LP filter has a cutoff frequency of 4 kHz when used
|
|
at a 44.1 kHz sample rate. Any FIR filter can be used
|
|
with the setAnalyzeConfig function.
|
|
</div>
|
|
<h3>Boards Supported</h3>
|
|
<ul>
|
|
<li>Teensy 3.2
|
|
<li>Teensy 3.5
|
|
<li>Teensy 3.6
|
|
<li>Teensy 4.0
|
|
<li>Teensy 4.1
|
|
</ul>
|
|
<h3>Audio Connections</h3>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=top><th>Port</th><th>Purpose</th></tr>
|
|
<tr class=odd><td align=center>In 0</td><td>Reference Sinewave</td></tr>
|
|
<tr class=odd><td align=center>In 1</td><td>Measure Sinewave</td></tr>
|
|
<tr class=odd><td align=center>Out 0</td><td>Measured Phase Angle</td></tr>
|
|
</table>
|
|
<h3>Functions</h3>
|
|
<p class=func><span class=keyword>setAnalyzePhaseConfig</span>(<strong>uint16_t</strong> LPType, <strong>float32_t</strong> *pCoeffs, <strong>uint16_t</strong> nCoeffs);</p>
|
|
<p class=desc> </p>
|
|
<p>Changes the output filter from the IIR default, where:
|
|
<ul>
|
|
<li>LPType is NO_LP_FILTER, IIR_LP_FILTER, FIR_LP_FILTER to select the output filter</li>
|
|
<li>pCoeffs is a pointer to filter coefficients, either IIR or FIR</li>
|
|
<li>nCoeffs is the number of filter coefficients</li>
|
|
</ul>
|
|
</p>
|
|
<p class=func><span class=keyword>setAnalyzePhaseConfig</span>(<strong>uint16_t</strong> LPType, <strong>float32_t</strong> *pCoeffs, <strong>uint16_t</strong> nCoeffs, <strong>uint16_t</strong> pdConfig);</p>
|
|
<p class=desc>Is the same as above, but controls pdConfig which is a bitwiseconfiguration selection selection:</li>
|
|
<pre class="desc">
|
|
Bit 0: 0=No Limiter (default) 1=Use limiter
|
|
Bit 2,1: 00=Use no acos linearizer
|
|
01=undefined
|
|
10=Fast, math-continuous acos() (default)
|
|
11=Accurate acosf()
|
|
Bit 3: 0=No scale of multiplier 1=scale to min-max (default)
|
|
Bit 4: 0=Output in degrees 1=Output in radians (default)
|
|
</pre>
|
|
</p>
|
|
<p class=func><span class=keyword>showError</span>(<strong>uint16_t</strong> e);</p>
|
|
<p class=desc>sets whether error printing comes from update (e=1) or not (e=0)
|
|
</p>
|
|
<p>This class supports programmable block size and sample rate via Settings_F32.</p>
|
|
|
|
<h3>Examples</h3>
|
|
<p class=exam>File > Examples > OpenAudio_ArduinoLibrary > AudioTestAnalyzePhase
|
|
</p>
|
|
<p class=exam>File > Examples > OpenAudio_ArduinoLibrary > AudioTestSinCos
|
|
</p>
|
|
|
|
<h3>Notes</h3>
|
|
<p>There are two inputs, 0 and 1 (Left and Right)
|
|
There is one output, the phase angle between 0 & 1 expressed in
|
|
radians (180 degrees is Pi radians) or degrees. This is a 180-degree
|
|
type of phase detector. See RadioIQMixer_F32 for a 360 degree type.
|
|
</p>
|
|
|
|
<p><strong>Defaults:</strong> 100 Hz IIR LP, output is in radians, and this does not need a call to begin().</p>
|
|
|
|
<p> A FIR LP is available where linear phase is needed, or NO_LP_FILTER that leaves
|
|
harmonics of the input frequency. A limiter is available to remove amplitude effects. Scaling of the output
|
|
levels can be done. Many more details are available from AudioAnalyzePhase_F32.h in
|
|
the OpenAudio_ArduinoLibrary. </p>
|
|
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioAnalyzePhase_F32">>
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
|
|
<div> <!-- Audio Input category -->
|
|
<script type="text/x-red" data-help-name="AudioEffectNoiseGate_F32">
|
|
<!-- ============ AudioEffectNoiseGate_F32 ======= -->
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>Apply envelope controlled threshold to a signal.
|
|
Inputs below a given level are removed.</p>
|
|
</div>
|
|
<h3>Boards Supported</h3>
|
|
<ul>
|
|
<li>Teensy 3.2
|
|
<li>Teensy 3.5
|
|
<li>Teensy 3.6
|
|
<li>Teensy 4.0
|
|
<li>Teensy 4.1
|
|
</ul>
|
|
<h3>Audio Connections</h3>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=top><th>Port</th><th>Purpose</th></tr>
|
|
<tr class=odd><td align=center>In 0</td><td>Signal In</td></tr>
|
|
<tr class=odd><td align=center>Out 0</td><td>Gated Signal Out</td></tr>
|
|
</table>
|
|
<h3>Functions</h3>
|
|
|
|
<p class=func><span class=keyword>setThreshold</span>(<strong>float </strong>dBFS);</p>
|
|
<p class=desc>Sets threshold to mute below, in dBFS (relative to full scale input),
|
|
such as -40.0.</p>
|
|
|
|
<p class=func><span class=keyword>setOpeningTime</span>(<strong>float </strong>tOpen);</p>
|
|
<p class=desc>In units of seconds, such as 0.02.</p>
|
|
|
|
<p class=func><span class=keyword>setClosingTime</span>(<strong>float </strong>tClose);</p>
|
|
<p class=desc>In units of seconds, such as 0.05.</p>
|
|
|
|
<p class=func><span class=keyword>setHoldTime</span>(<strong>float </strong>tHold);</p>
|
|
<p class=desc>In units of seconds, such as 0.10.</p>
|
|
|
|
<h3>Examples</h3>
|
|
<p class=exam>File > Examples > OpenAudio_ArduinoLibrary > NoiseGate
|
|
</p>
|
|
|
|
<h3>Notes</h3>
|
|
<p>Created: Max Huster, Feb 2021 </p>
|
|
<p>Purpose: This module mutes the Audio completly, when it's below a given threshold.</p>
|
|
<p>This processes a single stream fo audio data (i.e., it is mono)</p>
|
|
</script>
|
|
|
|
<script type="text/x-red" data-template-name="_AudioEffectNoiseGate_F32">
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
|
|
<div> <!-- Audio Input category -->
|
|
<script type="text/x-red" data-help-name="AudioEffectCompressor2_F32">
|
|
<!-- ============ AudioEffectCompressor2_F32 ======= -->
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>Apply envelope controlled gain to a signal.
|
|
Up to five independent "k dB per dB" segments. Includes expansion
|
|
(squelch) compression (AGC) and limiting.</p>
|
|
</div>
|
|
<h3>Boards Supported</h3>
|
|
<ul>
|
|
<li>Teensy 3.2
|
|
<li>Teensy 3.5
|
|
<li>Teensy 3.6
|
|
<li>Teensy 4.0
|
|
<li>Teensy 4.1
|
|
</ul>
|
|
<h3>Audio Connections</h3>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=top><th>Port</th><th>Purpose</th></tr>
|
|
<tr class=odd><td align=center>In 0</td><td>Signal In</td></tr>
|
|
<tr class=odd><td align=center>Out 0</td><td>Signal Out</td></tr>
|
|
</table>
|
|
<h3>Functions</h3>
|
|
|
|
<p class=func><span class=keyword>begin</span>();</p>
|
|
<p class=desc>Enables the compressor using parameters set
|
|
by the functon setCompressionCurve().
|
|
</p>
|
|
|
|
<p class=func><span class=keyword>setCompressionCurve</span>(<strong>struct</strong> compressionCurve*);</p>
|
|
<p class=desc>This defines the compression curve and needs an INO defined structure of type
|
|
compressionCurve. That structure is defined in AudioEffectCompressor2_F32.h.
|
|
</p> <!-- ADD INFO HERE <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< -->
|
|
|
|
<p class=func><span class=keyword>getCurrentInputDB</span>();</p>
|
|
<p class=desc>Returns the last input level in <strong>float</strong> dBFS.
|
|
</p>
|
|
|
|
<p class=func><span class=keyword>getvInMaxDB</span>();</p>
|
|
<p class=desc>Returns the maximum input level in <strong>float</strong> dBFS.
|
|
This is reset upon reading.
|
|
</p>
|
|
|
|
<p class=func><span class=keyword>getCurrentGainDB</span>();</p>
|
|
<p class=desc>Returns the current gain in <strong>float</strong> dB.
|
|
</p>
|
|
|
|
<p class=func><span class=keyword>setDelayBufferSize</span>(<strong>int16</strong> delay);</p>
|
|
<p class=desc>The input envelope detector can run ahead of the input
|
|
signal by use of this delay. This give improved transient response. The
|
|
parameter "delay" can be any power of 2 such as 64, 128 or 256.
|
|
A delay of 256 samples is 256/44100 = 0.0058 sec = 5.8 mSec.
|
|
</p>
|
|
|
|
<h3>Macros</h3>
|
|
<p>These are macros and not functions. They simplify getting a working compressor
|
|
and provide their own compressionCurve structure and are called with
|
|
a pointer to the compressor object (not to the structure) named "pobject" here.
|
|
This means the macro is invoked
|
|
without a preceeding object name. These macros replace the begin() function.
|
|
See the example program testCompressor2.ino below.</p>
|
|
|
|
<p class=func><span class=keyword>basicCompressorBegin</span>(pobject, linearInDB, compressionRatio);</p>
|
|
<p class=desc>The compression curve for basicCompressorBegin() has a 3 segments.
|
|
It is linear up to an input linearInDB
|
|
and then decreases gain according to compressionRatioDB up to an input -10 dB where it
|
|
is almost limited, with an increase of output level of 1 dB for a 10 dB increase
|
|
in input level. The output level at full input is 1 dB below full output.</p>
|
|
|
|
<p class=func><span class=keyword>limiterBegin</span>(pointerObject, float marDB, float linearInDB);</p>
|
|
<p class=desc>The compression curve for limiterBegin() has a 2 segments.
|
|
It is linear up to an input of linearInDB (typically -15.0f) and
|
|
then virtually limits for higher input levels. The output level at
|
|
this point is marDB, the margin to prevent clipping, like -2 dB.
|
|
This is not a clipper with waveform distortion, but rather decreases
|
|
the gain, dB for dB, as the input increases in the limiter region.</p>
|
|
|
|
<p class=func><span class=keyword>squelchCompressorBegin</span>(pobject, squelchInDB, linearInDB, compressionInDB, compressionRatio);</p>
|
|
<p class=desc>The compression curve for squelchCompressorBegin() has a 3 segments.
|
|
and is similar to basicCompression above, except that there is
|
|
an expansion region for low levels. So, the call defines the four regions in
|
|
terms of the input levels. squelchInDB sets the lowest input level
|
|
before the squelching effect starts.</p>
|
|
|
|
<h3>Examples</h3>
|
|
<p class=exam>File > Examples > OpenAudio_ArduinoLibrary > testCompressor2
|
|
</p>
|
|
|
|
<h3>Notes</h3>
|
|
<p>This is a general purpose audio compressor block (C++ class). It works by determining
|
|
the average input of the input signal, and based on a pre-determined curve,
|
|
changes the gain going through the block.
|
|
A good discussion is the Wikipedia page:
|
|
https://en.wikipedia.org/wiki/Dynamic_range_compression
|
|
This compressor includes up to 5 dB/dB line segments. allowing
|
|
considerable flexibility including:<ul>
|
|
<li> Multi segment compression curves, up to 5</li>
|
|
<li> Limiting</li>
|
|
<li> Approximation to "soft knees"</li>
|
|
<li> Expansion for suppressing low-level artifacts</li>
|
|
<li> Anticipation</li>
|
|
<li> Scale offset for use such as hearing-aid audiology</li>
|
|
</ul>
|
|
This is derived from the WDRC compressor. Chip Audette (OpenAudio) Feb 2017
|
|
Which was derived From: WDRC_circuit from CHAPRO from BTNRC: https://github.com/BTNRH/chapro</p>
|
|
</script>
|
|
|
|
<script type="text/x-red" data-template-name="_AudioCompressor2_F32">
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
|
|
<script type="text/x-red" data-help-name="AudioEffectEmpty_F32">
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>This module does nothing. It passes through the signal with no
|
|
changes. If you open up the code, you can copy it to use as a template
|
|
to build your own audio processing block.
|
|
</p>
|
|
</div>
|
|
<h3>Audio Connections</h3>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=top><th>Port</th><th>Purpose</th></tr>
|
|
<tr class=odd><td align=center>In 0</td><td>Input Signal</td></tr>
|
|
<tr class=odd><td align=center>Out 0</td><td>Output Signal</td></tr>
|
|
</table>
|
|
<h3>Functions</h3>
|
|
<p>Being an empty block, it has no functions.</p>
|
|
<!--
|
|
<h3>Examples</h3>
|
|
<p class=exam>File > Examples > Audio >
|
|
</p>
|
|
-->
|
|
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioEffectEmpty_F32">
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
<script type="text/x-red" data-help-name="AudioEffectGain_F32">
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>Scale the signal by a fixed amount. The gain is applied on a per-block
|
|
(as opposed to a per-sample) basis.
|
|
</p>
|
|
</div>
|
|
<h3>Audio Connections</h3>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=top><th>Port</th><th>Purpose</th></tr>
|
|
<tr class=odd><td align=center>In 0</td><td>Signal Input</td></tr>
|
|
<tr class=odd><td align=center>Out 0</td><td>Result of Input Times Gain</td></tr>
|
|
</table>
|
|
<h3>Functions</h3>
|
|
<p>There are two functions for setting the gain of this block:</p>
|
|
<p class=func><span class=keyword>setGain</span>(gain);</p>
|
|
<p class=desc>Set the gain as a linear value. A value between 0.0 and 1.0
|
|
attenuates the signal while a value above 1.0 amplifies the signal. Negative
|
|
values will invert the signal.</p>
|
|
<p class=func><span class=keyword>setGain_dB</span>(gain_dB);</p>
|
|
<p class=desc>Set the gain using a decibel value (gain_dB = 20*log10(gain)).
|
|
A value less than zero attenuates the signal and a value greater than zero
|
|
amplifies the signal. Phase inversion is not available.</p>
|
|
<!--
|
|
<h3>Examples</h3>
|
|
<p class=exam>File > Examples > Audio >
|
|
</p>
|
|
-->
|
|
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioEffectGain_F32">
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
|
|
<script type="text/x-red" data-help-name="AudioFilterBiquad_F32">
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>Biquadratic cascaded filter, useful for all sorts of filtering.
|
|
Up to 4 stages may be cascaded.
|
|
</p>
|
|
<p align=center><img src="img/biquad.png"></p>
|
|
</div>
|
|
<h3>Audio Connections</h3>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=top><th>Port</th><th>Purpose</th></tr>
|
|
<tr class=odd><td align=center>In 0</td><td>Signal to be filtered</td></tr>
|
|
<tr class=odd><td align=center>Out 0</td><td>Filtered Signal Output</td></tr>
|
|
</table>
|
|
<h3>Functions</h3>
|
|
<p class=func><span class=keyword>setLowpass</span>(stage, frequency, Q);</p>
|
|
<p class=desc>Configure one stage of the filter (0 to 3) with low pass
|
|
response, with the specified corner frequency and Q shape. If Q is
|
|
higher that 0.7071, be careful of filter gain (see below).
|
|
</p>
|
|
<p class=func><span class=keyword>setHighpass</span>(stage, frequency, Q);</p>
|
|
<p class=desc>Configure one stage of the filter (0 to 3) with high pass
|
|
response, with the specified corner frequency and Q shape. If Q is
|
|
higher that 0.7071, be careful of filter gain (see below).
|
|
</p>
|
|
<p class=func><span class=keyword>setBandpass</span>(stage, frequency, Q);</p>
|
|
<p class=desc>Configure one stage of the filter (0 to 3) with band pass
|
|
response. The filter has unity gain at the specified frequency. Q
|
|
controls the width of frequencies allowed to pass.
|
|
</p>
|
|
<p class=func><span class=keyword>setNotch</span>(stage, frequency, Q);</p>
|
|
<p class=desc>Configure one stage of the filter (0 to 3) with band reject (notch)
|
|
response. Q controls the width of rejected frequencies.
|
|
</p>
|
|
<p class=func><span class=keyword>setLowShelf</span>(stage, frequency, gain, slope);</p>
|
|
<p class=desc>Configure one stage of the filter (0 to 3) with low shelf response.
|
|
A low shelf filter attenuates or amplifies signals below the specified frequency.
|
|
Frequency controls the slope midpoint, gain is in dB and can be both
|
|
positive or negative. The slope parameter controls steepness of gain transition.
|
|
A slope of 1 yields maximum steepness without overshoot,
|
|
lower values yield a less steep slope. See the picture below for a visualization
|
|
of the slope parameter's effect.
|
|
Be careful with positive gains and slopes higher than 1 as they introduce gain
|
|
(see warning below).
|
|
</p>
|
|
</p>
|
|
<p class=func><span class=keyword>setHighShelf</span>(stage, frequency, gain, slope);</p>
|
|
<p class=desc>Configure one stage of the filter (0 to 3) with high shelf response.
|
|
A high shelf filter attenuates or amplifies signals above the specified frequency.
|
|
Frequency controls the slope midpoint, gain is in dB and can be both
|
|
positive or negative. The slope parameter controls steepness of gain transition.
|
|
A slope of 1 yields maximum steepness without overshoot,
|
|
lower values yield a less steep slope. See the picture below for a visualization
|
|
of the slope parameter's effect.
|
|
Be careful with positive gains and slopes higher than 1 as they introduce gain
|
|
(see warning below).
|
|
</p>
|
|
<p align=center><img src="img/shelf_filter.png"></p>
|
|
<p class=func><span class=keyword>setCoefficients</span>(stage, array[5]);</p>
|
|
<p class=desc>Configure one stage of the filter (0 to 3) with an arbitrary
|
|
filter response. The array of coefficients is in order: B0, B1, B2, A1, A2.
|
|
Each coefficient must be less than 2.0 and greater than -2.0. The array
|
|
should be type double. Alternately, it may be type int, where 1.0 is
|
|
represented with 1073741824 (2<sup>30</sup>).
|
|
</p>
|
|
<p class=func><span class=keyword><strong>double* </strong>getCoefficients</span>();</p>
|
|
<p class=desc>Returns a pointer to the array of double precision coefficients. For
|
|
up to four stages, each stage is arranged in order B0, B1, B2, A1, A2. This is a
|
|
maximum of 20 coefficients with unused stages showing a 1.0 and four zeros. </p>
|
|
<h3>Examples</h3>
|
|
<p class=exam>File > Examples > Audio > Effects > Filter
|
|
</p>
|
|
<h3>Notes</h3>
|
|
<p>Filters can with gain must have their input signals attenuated, so the
|
|
signal does not exceed 1.0.
|
|
</p>
|
|
<p>This object implements up to 4 cascaded stages. Unconfigured stages will
|
|
not pass any signal.
|
|
</p>
|
|
<p>Biquad filters with low corner frequency (under about 400 Hz) can run into
|
|
trouble with limited numerical precision, causing the filter to perform
|
|
poorly. For very low corner frequency, the State Variable (Chamberlin)
|
|
filter should be used.
|
|
</p>
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioFilterBiquad_F32">
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
<script type="text/x-red" data-help-name="AudioFilterFIR_F32">
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>Finite impulse response filter, useful for all sorts of filtering.
|
|
</p>
|
|
<p align=center><img src="img/fir_filter.png"></p>
|
|
</div>
|
|
<h3>Audio Connections</h3>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=top><th>Port</th><th>Purpose</th></tr>
|
|
<tr class=odd><td align=center>In 0</td><td>Signal to be filtered</td></tr>
|
|
<tr class=odd><td align=center>Out 0</td><td>Filtered Signal Output</td></tr>
|
|
</table>
|
|
<h3>Functions</h3>
|
|
<p class=func><span class=keyword>begin</span>(filter_coeff, filter_length, block_size);</p>
|
|
<p class=desc>Initialize the filter. The filter_coeff must be an array of 32-bit floats (the
|
|
filter's impulse response), the filter_length indicates the number of points in the array,
|
|
and block_size is the length of the audio block that will be passed to this filtering
|
|
object during operation. The filter_coeff array may also be set as
|
|
FIR_PASSTHRU (with filter_length = 0), to directly pass the input to output without
|
|
filtering.
|
|
</p>
|
|
<p class=func><span class=keyword>end</span>();</p>
|
|
<p class=desc>Turn the filter off.
|
|
</p>
|
|
<!--
|
|
<h3>Examples</h3>
|
|
<p class=exam>File > Examples > Audio > Effects > Filter_FIR
|
|
</p>
|
|
-->
|
|
<h3>Known Issues</h3>
|
|
<p>Your filter's impulse response array must have an even length. If you have
|
|
add odd number of taps, you must add an extra zero to increase the length
|
|
to an even number.
|
|
</p>
|
|
<p>The minimum number of taps is 4. If you use less, add extra zeros to increase
|
|
the length to 4.
|
|
</p>
|
|
<p>The impulse response must be given in reverse order. Many filters have
|
|
symetrical impluse response, making this a non-issue. If your filter has
|
|
a non-symetrical response, make sure the data is in reverse time order.
|
|
</p>
|
|
<h3>Notes</h3>
|
|
<p>FIR filters requires more CPU time than Biquad (IIR), but they can
|
|
implement filters with better phase response.
|
|
</p>
|
|
<p>The free
|
|
<a href="http://t-filter.engineerjs.com/" target="_blank"> TFilter Design Tool</a>
|
|
can be used to create the impulse response array. Be sure to choose the desired sampling
|
|
frequency (that tool defaults to only 2000 Hz whereas Teensy defaults to 44117) and
|
|
for this library the output type should be "float" (32 bit).
|
|
</p>
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioFilterFIR_F32">
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
<script type="text/x-red" data-help-name="AudioFilterFreqWeighting_F32">
|
|
<p>Frequency weighting. Defaults to A-Weighting</p>
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioFilterFreqWeighting_F32 ">
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
<script type="text/x-red" data-help-name="AudioFilterTimeWeighting_F32">
|
|
<p>Time weighting for sound level meter. Defaults to SLOW</p>
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioFilterTimeWeighting_F32 ">
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
<script type="text/x-red" data-help-name="AudioMathAdd_F32">
|
|
<p> AudioMathAdd_F32</p>
|
|
<p> Created: Chip Audette, Open Audio, July 2018</p>
|
|
<p> Purpose: Add together two channels (vectors) of audio data on a point-by-point basis</p>
|
|
<p> (like AudioMathMutiply, but addition). Assumes floating-point data.</p>
|
|
<p> This processes a single stream fo audio data (ie, it is mono)</p>
|
|
<p> MIT License. use at your own risk.</p>
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioMathAdd_F32 ">
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
<script type="text/x-red" data-help-name="AudioMathMultiply_F32">
|
|
<p> AudioMathMultiply</p>
|
|
<p> Created: Patrick Radius, December 2016</p>
|
|
<p> Purpose: Multiply two channels of audio data. Can be used for example as 'vca' or amplitude modulation.</p>
|
|
<p> Assumes floating-point data.</p>
|
|
<p> This processes a single stream fo audio data (ie, it is mono)</p>
|
|
<p> MIT License. use at your own risk.</p>
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioMathMultiply_F32 ">
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
<script type="text/x-red" data-help-name="AudioMathOffset_F32">
|
|
<p> AudioMathOffset_F32</p>
|
|
<p> Created: Chip Audette, Open Audio, June 2018</p>
|
|
<p> Purpose: Add a constant DC value to all audio samples</p>
|
|
<p> Assumes floating-point data.</p>
|
|
<p> This processes a single stream fo audio data (ie, it is mono)</p>
|
|
<p> MIT License. use at your own risk.</p>
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioMathOffset_F32 ">
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
<script type="text/x-red" data-help-name="AudioMathScale_F32">
|
|
<p> AudioMathScale_F32</p>
|
|
<p> Created: Chip Audette, Open Audio, June 2018</p>
|
|
<p> Purpose: Multiply all audio samples by a single value (not vector)</p>
|
|
<p> Assumes floating-point data.</p>
|
|
<p> This processes a single stream fo audio data (ie, it is mono)</p>
|
|
<p> MIT License. use at your own risk.</p>
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioMathScale_F32 ">
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
<script type="text/x-red" data-help-name="AudioMixer4_F32">
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>Combine up to 4 audio signals together, each with adjustable gain.
|
|
All channels support signal attenuation or amplification.</p>
|
|
</div>
|
|
<h3>Audio Connections</h3>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=top><th>Port</th><th>Purpose</th></tr>
|
|
<tr class=odd><td align=center>In 0</td><td>Input signal #1</td></tr>
|
|
<tr class=odd><td align=center>In 1</td><td>Input signal #2</td></tr>
|
|
<tr class=odd><td align=center>In 2</td><td>Input signal #3</td></tr>
|
|
<tr class=odd><td align=center>In 3</td><td>Input signal #4</td></tr>
|
|
<tr class=odd><td align=center>Out 0</td><td>Sum of all inputs</td></tr>
|
|
</table>
|
|
<h3>Functions</h3>
|
|
<p class=func><span class=keyword>gain</span>(channel, level);</p>
|
|
<p class=desc>Adjust the amplification or attenuation. "channel" must
|
|
be 0 to 3. "level" may be any floating point number. A level value of
|
|
1.0 passes the signal through directly. Between 0 to 1.0 attenuates the signal, and above
|
|
1.0 amplifies it. A level value of 0 shuts the channel
|
|
off completely. All 4 channels have separate settings.
|
|
</p>
|
|
<!--
|
|
<h3>Examples</h3>
|
|
<p class=exam>File > Examples > Audio > SamplePlayer
|
|
</p>
|
|
<p class=exam>File > Examples > Audio > Synthesis > PlaySynthMusic
|
|
</p>
|
|
<p class=exam>File > Examples > Audio > Analysis > SpectrumAnalyzerBasic
|
|
</p>
|
|
<p class=exam>File > Examples > Audio > Analysis > DialTone_Serial
|
|
</p>
|
|
<p class=exam>File > Examples > Audio > MemoryAndCpuUsage
|
|
</p>
|
|
-->
|
|
<h3>Notes</h3>
|
|
<p>More than 4 channels may be combined by connecting multiple mixers
|
|
in tandem. For example, a 16 channel mixer may be built using 5
|
|
mixers, where the fifth mixer combines the outputs of the first 4.
|
|
</p>
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioMixer4_F32">
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
<script type="text/x-red" data-help-name="AudioMixer8_F32">
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>Combine up to 8 audio signals together, each with adjustable gain.
|
|
All channels support signal attenuation or amplification.</p>
|
|
</div>
|
|
<h3>Audio Connections</h3>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=top><th>Port</th><th>Purpose</th></tr>
|
|
<tr class=odd><td align=center>In 0</td><td>Input signal #1</td></tr>
|
|
<tr class=odd><td align=center>In 1</td><td>Input signal #2</td></tr>
|
|
<tr class=odd><td align=center>In 2</td><td>Input signal #3</td></tr>
|
|
<tr class=odd><td align=center>In 3</td><td>Input signal #4</td></tr>
|
|
<tr class=odd><td align=center>In 4</td><td>Input signal #5</td></tr>
|
|
<tr class=odd><td align=center>In 5</td><td>Input signal #6</td></tr>
|
|
<tr class=odd><td align=center>In 6</td><td>Input signal #7</td></tr>
|
|
<tr class=odd><td align=center>In 7</td><td>Input signal #8</td></tr>
|
|
<tr class=odd><td align=center>Out 0</td><td>Sum of all inputs</td></tr>
|
|
</table>
|
|
<h3>Functions</h3>
|
|
<p class=func><span class=keyword>gain</span>(channel, level);</p>
|
|
<p class=desc>Adjust the amplification or attenuation. "channel" must
|
|
be 0 to 7. "level" may be any floating point number. A level value of
|
|
1.0 passes the signal through directly. Between 0 to 1.0 attenuates the signal, and above
|
|
1.0 amplifies it. A level value of 0 shuts the channel
|
|
off completely. All 4 channels have separate settings.
|
|
</p>
|
|
<!--
|
|
<h3>Examples</h3>
|
|
<p class=exam>File > Examples > Audio > SamplePlayer
|
|
</p>
|
|
<p class=exam>File > Examples > Audio > Synthesis > PlaySynthMusic
|
|
</p>
|
|
<p class=exam>File > Examples > Audio > Analysis > SpectrumAnalyzerBasic
|
|
</p>
|
|
<p class=exam>File > Examples > Audio > Analysis > DialTone_Serial
|
|
</p>
|
|
<p class=exam>File > Examples > Audio > MemoryAndCpuUsage
|
|
</p>
|
|
-->
|
|
<h3>Notes</h3>
|
|
<p>More than 8 channels may be combined by connecting multiple mixers
|
|
in tandem. For example, a 16 channel mixer may be built using three
|
|
mixers, where the third mixer combines the outputs of the first two.
|
|
</p>
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioMixer8_F32">
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
<script type="text/x-red" data-help-name="FFT_Overlapped_F32">
|
|
<p> FFT_Overrlapped_F32</p>
|
|
<p> Purpose: Encapsulate the ARM floating point FFT/IFFT functions</p>
|
|
<p> in a way that naturally interfaces to my float32</p>
|
|
<p> extension of the Teensy Audio Library.</p>
|
|
<p> Provides functionality to do overlapped FFT/IFFT where</p>
|
|
<p> each audio block is a fraction (1, 1/2, 1/4) of the</p>
|
|
<p> totaly FFT length. This class handles all of the</p>
|
|
<p> data shuffling to composite the previous data blocks</p>
|
|
<p> with the current data block to provide the full FFT.</p>
|
|
<p> Does similar data shuffling (overlapp-add) for IFFT.</p>
|
|
<p> Created: Chip Audette (openaudio.blogspot.com)</p>
|
|
<p> Jan-Jul 2017</p>
|
|
<p> Typical Usage as FFT:</p>
|
|
<p> //setup the audio stuff</p>
|
|
<p> float sample_rate_Hz = 44100.0; //define sample rate</p>
|
|
<p> int audio_block_samples = 32; //define size of audio blocks</p>
|
|
<p> AudioSettings_F32 audio_settings(sample_rate_Hz, audio_block_samples);</p>
|
|
<p> // ... continue creating all of your Audio Processing Blocks ...</p>
|
|
<p> // within a custom audio processing algorithm that you've written</p>
|
|
<p> // you'd create the FFT and IFFT elements</p>
|
|
<p> int NFFT = 128; //define length of FFT that you want (multiple of audio_block_samples)</p>
|
|
<p> FFT_Overrlapped_F32 FFT_obj(audio_settings,NFFT); //Creare FFT object</p>
|
|
<p> FFT_Overrlapped_F32 IFFT_obj(audio_settings,NFFT); //Creare IFFT object</p>
|
|
<p> float complex_2N_buffer[2*NFFT]; //create buffer to hold the FFT output</p>
|
|
<p> // within your own algorithm's "update()" function (which is what</p>
|
|
<p> // is called automatically by the Teensy Audio Libarary approach</p>
|
|
<p> // to audio processing), you can execute the FFT and IFFT</p>
|
|
<p> // First, get the audio and convert to frequency-domain using an FFT</p>
|
|
<p> audio_block_f32_t *in_audio_block = AudioStream_F32::receiveReadOnly_f32();</p>
|
|
<p> FFT_obj.execute(in_audio_block, complex_2N_buffer); //output is in complex_2N_buffer</p>
|
|
<p> AudioStream_F32::release(in_audio_block); //We just passed ownership to FFT_obj, so release it here.</p>
|
|
<p> // Next do whatever processing you'd like on the frequency domain data</p>
|
|
<p> // that is held in complex_2N_buffer</p>
|
|
<p> // Finally, you can convert back to the time domain via IFFT</p>
|
|
<p> audio_block_f32_t *out_audio_block = IFFT_obj.execute(complex_2N_buffer);</p>
|
|
<p> //note that the "out_audio_block" is mananged by IFFT_obj, so don't worry about releasing it.</p>
|
|
<p> License: MIT License</p>
|
|
</script>
|
|
<script type="text/x-red" data-template-name="FFT_Overlapped_F32 ">
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
<script type="text/x-red" data-help-name="IFFT_Overlapped_F32">
|
|
<p> FFT_Overrlapped_F32</p>
|
|
<p> Purpose: Encapsulate the ARM floating point FFT/IFFT functions</p>
|
|
<p> in a way that naturally interfaces to my float32</p>
|
|
<p> extension of the Teensy Audio Library.</p>
|
|
<p> Provides functionality to do overlapped FFT/IFFT where</p>
|
|
<p> each audio block is a fraction (1, 1/2, 1/4) of the</p>
|
|
<p> totaly FFT length. This class handles all of the</p>
|
|
<p> data shuffling to composite the previous data blocks</p>
|
|
<p> with the current data block to provide the full FFT.</p>
|
|
<p> Does similar data shuffling (overlapp-add) for IFFT.</p>
|
|
<p> Created: Chip Audette (openaudio.blogspot.com)</p>
|
|
<p> Jan-Jul 2017</p>
|
|
<p> Typical Usage as FFT:</p>
|
|
<p> //setup the audio stuff</p>
|
|
<p> float sample_rate_Hz = 44100.0; //define sample rate</p>
|
|
<p> int audio_block_samples = 32; //define size of audio blocks</p>
|
|
<p> AudioSettings_F32 audio_settings(sample_rate_Hz, audio_block_samples);</p>
|
|
<p> // ... continue creating all of your Audio Processing Blocks ...</p>
|
|
<p> // within a custom audio processing algorithm that you've written</p>
|
|
<p> // you'd create the FFT and IFFT elements</p>
|
|
<p> int NFFT = 128; //define length of FFT that you want (multiple of audio_block_samples)</p>
|
|
<p> FFT_Overrlapped_F32 FFT_obj(audio_settings,NFFT); //Creare FFT object</p>
|
|
<p> FFT_Overrlapped_F32 IFFT_obj(audio_settings,NFFT); //Creare IFFT object</p>
|
|
<p> float complex_2N_buffer[2*NFFT]; //create buffer to hold the FFT output</p>
|
|
<p> // within your own algorithm's "update()" function (which is what</p>
|
|
<p> // is called automatically by the Teensy Audio Libarary approach</p>
|
|
<p> // to audio processing), you can execute the FFT and IFFT</p>
|
|
<p> // First, get the audio and convert to frequency-domain using an FFT</p>
|
|
<p> audio_block_f32_t *in_audio_block = AudioStream_F32::receiveReadOnly_f32();</p>
|
|
<p> FFT_obj.execute(in_audio_block, complex_2N_buffer); //output is in complex_2N_buffer</p>
|
|
<p> AudioStream_F32::release(in_audio_block); //We just passed ownership to FFT_obj, so release it here.</p>
|
|
<p> // Next do whatever processing you'd like on the frequency domain data</p>
|
|
<p> // that is held in complex_2N_buffer</p>
|
|
<p> // Finally, you can convert back to the time domain via IFFT</p>
|
|
<p> audio_block_f32_t *out_audio_block = IFFT_obj.execute(complex_2N_buffer);</p>
|
|
<p> //note that the "out_audio_block" is mananged by IFFT_obj, so don't worry about releasing it.</p>
|
|
<p> License: MIT License</p>
|
|
</script>
|
|
<script type="text/x-red" data-template-name="IFFT_Overlapped_F32 ">
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
<script type="text/x-red" data-help-name="AudioInputI2S_F32">
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>Receive float32 stereo audio from the
|
|
<a href="http://www.pjrc.com/store/teensy3_audio.html" target="_blank">Teensy Audio Board</a>,
|
|
the <a href="https://www.tympan.org/">Tympan Audio Board</a>,
|
|
or another I2S device.</p>
|
|
</div>
|
|
<h3>Audio Connections</h3>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=top><th>Port</th><th>Purpose</th></tr>
|
|
<tr class=odd><td align=center>Out 0</td><td>Left Channel</td></tr>
|
|
<tr class=odd><td align=center>Out 1</td><td>Right Channel</td></tr>
|
|
</table>
|
|
<h3>Constructor</h3>
|
|
<p>Objects of this class can be created using the typical, minimal constructor using no arguments.
|
|
Or, to control the sample rate and block size, the constructor can be passed an AudioSettings_F32
|
|
object that has been configured with your desired settings.</p>
|
|
<h3>Functions</h3>
|
|
<p>This object has no functions to call from the Arduino sketch. It
|
|
simply streams data from the I2S hardware to its 2 output ports.</p>
|
|
<h3>Hardware</h3>
|
|
<p>The I2S signals are used in "master" mode, where the Teensy creates
|
|
all 3 clock signals and controls all data timing.</p>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=top><th>Pin</th><th>Signal</th><th>Direction</th></tr>
|
|
<tr class=odd><td align=center>9</td><td>BCLK</td><td>Output</td></tr>
|
|
<tr class=odd><td align=center>11</td><td>MCLK</td><td>Output</td></tr>
|
|
<tr class=odd><td align=center>22</td><td>TX</td><td>Output</td></tr>
|
|
<tr class=odd><td align=center>23</td><td>LRCLK</td><td>Output</td></tr>
|
|
</table>
|
|
<p>Audio from
|
|
master mode I2S may be used in the same project as ADC, DAC and
|
|
PWM signals, because all remain in sync to Teensy's timing</p>
|
|
<!--
|
|
<h3>Examples</h3>
|
|
<p class=exam>File > Examples > Audio > HardwareTesting > PassThroughStereo
|
|
</p>
|
|
<p class=exam>File > Examples > Audio > Recorder
|
|
</p>
|
|
<p class=exam>File > Examples > Audio > Analysis > PeakMeterStereo
|
|
</p>
|
|
<p class=exam>File > Examples > Audio > Analysis > FFT
|
|
</p>
|
|
<p class=exam>File > Examples > Audio > Analysis > SpectrumAnalyzerBasic
|
|
</p>
|
|
<p class=exam>File > Examples > Audio > Effects > Chorus
|
|
</p>
|
|
<p class=exam>File > Examples > Audio > Effects > Flange
|
|
</p>
|
|
<p class=exam>File > Examples > Audio > Effects > Filter
|
|
</p>
|
|
<p class=exam>File > Examples > Audio > Effects > Filter_FIR
|
|
</p>
|
|
-->
|
|
<h3>Notes</h3>
|
|
<p>Only one I2S input and one I2S output object may be used. Master
|
|
and slave modes may not be mixed (both must be of the same type).
|
|
</p>
|
|
<p>Being an "_F32" object, audio is passed into this class using F32 memory
|
|
that was allocated using an AudioMemory_F32() call in your sketch's setup()
|
|
routine. But, internally, this class still uses some Int16 data handling,
|
|
so be sure to also include an AudioMemory() call in addition to AudioMemory_F32()
|
|
to allocate the Int16 memory.</p>
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioInputI2S_F32">
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
<script type="text/x-red" data-help-name="AudioOutputI2S_F32">
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>Transmit float32 stereo audio to the various I2S Codec boards, such as
|
|
<a href="http://www.pjrc.com/store/teensy3_audio.html" target="_blank">Teensy Audio Board</a>.
|
|
</div>
|
|
<h3>Audio Connections</h3>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=top><th>Port</th><th>Purpose</th></tr>
|
|
<tr class=odd><td align=center>In 0</td><td>Left Channel</td></tr>
|
|
<tr class=odd><td align=center>In 1</td><td>Right Channel</td></tr>
|
|
</table>
|
|
<h3>Constructor</h3>
|
|
<p>Objects of this class can be created using the typical, minimal constructor using no arguments.
|
|
Or, to control the sample rate and block size, the constructor can be passed an AudioSettings_F32
|
|
object that has been configured with your desired settings.</p>
|
|
<h3>Functions</h3>
|
|
<p class=func><span class=keyword>setGain</span>(<strong>float </strong>Gain);</p>
|
|
<p class=desc>Sets the output gain of both left and right channels. Values are unconstrained, but
|
|
normally range from 0.0 to 1.0 (default 1.0). A setting of exactly 1.0 results in
|
|
no scaling and saves a small amount of processor. </p>
|
|
<h3>Hardware</h3>
|
|
<p>A control Codec object, such as the SGTL5000 must be included.</p>
|
|
<p>Audio from
|
|
master mode I2S may be used in the same project as ADC, DAC (not T4.x) and
|
|
PWM signals, because all remain in sync to Teensy's timing</p>
|
|
<!--
|
|
<h3>Examples</h3>
|
|
<p>Nearly all the examples use this object. Here are some of the highlights:</p>
|
|
<p class=exam>File > Examples > Audio > HardwareTesting > PassThroughStereo
|
|
</p>
|
|
<p class=exam>File > Examples > Audio > SamplePlayer
|
|
</p>
|
|
<p class=exam>File > Examples > Audio > Recorder
|
|
</p>
|
|
<p class=exam>File > Examples > Audio > WavFilePlayer
|
|
</p>
|
|
<p class=exam>File > Examples > Audio > Effects > Chorus
|
|
</p>
|
|
<p class=exam>File > Examples > Audio > Synthesis > PlaySynthMusic
|
|
</p>
|
|
-->
|
|
<h3>Notes</h3>
|
|
<p>Only one I2S input and one I2S output object may be used. Master
|
|
and slave modes may not be mixed (both must be of the same type).</p>
|
|
<p>Being an "_F32" object, audio is passed into this class using F32 memory
|
|
that was allocated using an AudioMemory_F32() call in your sketch's setup()
|
|
routine. But, internally, this class still uses some Int16 data handling,
|
|
so be sure to also include an AudioMemory() call in addition to AudioMemory_F32()
|
|
to allocate the Int16 memory.</p>
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioOutputI2S_F32">
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
|
|
<script type="text/x-red" data-help-name="AudioAlignLR_F32">
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>Measure Codec L-R time misalignment at startup and make
|
|
time shift corrections during regular operation. ** BETA TEST **</p>
|
|
</div>
|
|
<h3>Boards Supported</h3>
|
|
<ul>
|
|
<li>Teensy 3.5
|
|
<li>Teensy 3.6
|
|
<li>Teensy 4.0
|
|
<li>Teensy 4.1
|
|
</ul>
|
|
|
|
<h3>Audio Connections</h3>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=top><th>Port</th><th>Purpose</th></tr>
|
|
<tr class=odd><td align=center>In 0</td><td>Left Channel</td></tr>
|
|
<tr class=odd><td align=center>In 1</td><td>Right Channel</td></tr>
|
|
<tr class=odd><td align=center>Out 0</td><td>Left Channel</td></tr>
|
|
<tr class=odd><td align=center>Out 1</td><td>Right Channel</td></tr>
|
|
<tr class=odd><td align=center>Out 2</td><td>Test Signal (Beta)</td></tr>
|
|
</table>
|
|
|
|
<h3>Functions</h3>
|
|
<p class=func><span class=keyword>read</span>(<strong>void</strong>);</p>
|
|
<p class=desc>This returns a pointer to a <strong>TPinfo</strong> structure
|
|
containing information on the operation of the time alignment. The structure
|
|
members are:</p>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=odd><td align=center><strong>uint16_t</strong></td><td> TPstate</td></tr>
|
|
<tr class=odd><td align=center><strong>uint32_t</strong></td><td> nMeas</td></tr>
|
|
<tr class=odd><td align=center><strong>float32_t</strong></td><td> xNorm</td></tr>
|
|
<tr class=odd><td align=center><strong>float32_t</strong></td><td> xcVal[4]</td></tr>
|
|
<tr class=odd><td align=center><strong>int16_t</strong></td><td> neededShift</td></tr>
|
|
<tr class=odd><td align=center><strong>int16_t</strong></td><td> TPerror</td></tr>
|
|
<tr class=odd><td align=center><strong>uint16_t</strong></td><td> TPsignalHardware</td></tr>
|
|
</table>
|
|
|
|
<p class=func><span class=keyword>stateTwinPeaks</span>(<strong>uint16_t</strong> TPstate);</p>
|
|
<p class=desc>This process comes up set to TP_IDLE. To start the alignment detection,
|
|
use this function to set TPstate to TP_MEASURE. If detection is successful,
|
|
the state will automatically move to TP_RUN. Correction of the alignment error will
|
|
then continue without requiring any function calls.</p>
|
|
|
|
<!--
|
|
<p class=func><span class=keyword>setThreshold</span>(<strong>float32_t</strong> TPthreshold);</p>
|
|
<p class=desc>The L-R cross correlation signal must exceed this threshold to indicate
|
|
detection. To set this value run the process with any
|
|
value for the threshold. Examine the three -1, 0, 1 output data around update #15.
|
|
Then set the threshold to about half of maximum positive value of the three.</p>
|
|
-->
|
|
|
|
<h3>Examples</h3>
|
|
<p class=exam>File > Examples > TestTwinPeaks
|
|
</p>
|
|
<h3>Notes</h3>
|
|
<p>The object construction takes parameters, the first three being
|
|
required:
|
|
</p>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=odd><td align=center><strong>uint16_t</strong></td><td>Hardware</td></tr>
|
|
<tr class=odd><td align=center><strong>uint16_t</strong></td><td>Pin</td></tr>
|
|
<tr class=odd><td align=center><strong>bool</strong></td><td>Invert</td></tr>
|
|
</table>
|
|
<p>See AudioAlignLR.h and the example INO for details on the parameters.
|
|
A fourth parameter, settings, is optional.</p>
|
|
|
|
<p>There are two different methods for generating a test signal. Both
|
|
require a small amount of hardware to be contructed. The first, and recommended,
|
|
(Hardware=TP_SIGNAL_IO_PIN) method uses a dedicated Teensy digital I/O pin to generate an 11.4 kHz square wave that
|
|
is permanently connected by two 10K to 100K resistors to the L&R ADC inputs. The second,
|
|
(Hardware=TP_SIGNAL_CODEC), method briefly borrows the
|
|
Codec R channel output to produce an 11 kHz square wave that is coupled via analog
|
|
switches to the L&R ADC inputs. Both methods seem to work very well. The first method's
|
|
hardware is simpler. The third audio output from the AlignLR block is not used for
|
|
the first method.</p>
|
|
|
|
<p>The resistor values, referred to above, depend on the impedance that
|
|
the ADC is being driven from. As a general rule, they should be (very) roughly 200
|
|
times the driving point impedance. Too high a value reults in noise in the
|
|
measurement. Too low can cause digital noise to get into the audio. It is
|
|
also possible to use a pair of NPN transistors, such as a 2N3904, to shunt the resistor
|
|
paths to the Codecanalog ground. For that case the resistor would be divided into two
|
|
parts, the sum of which equaled the desired resistor value.</p>
|
|
|
|
<p>To DO: 1 - See if anybody needs the Codec DAC method.
|
|
Remove if not needed.</p>
|
|
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioAlignLR_F32">
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
<script type="text/x-red" data-help-name="AudioInputUSB_F32">
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>Receive stereo audio from a PC or Mac. Teensy appears as a USB
|
|
sound device.</p>
|
|
<p align=center><img src="img/usbtype_audio_in.png"></p>
|
|
</div>
|
|
<h3>Audio Connections</h3>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=top><th>Port</th><th>Purpose</th></tr>
|
|
<tr class=odd><td align=center>Out 0</td><td>Left Channel</td></tr>
|
|
<tr class=odd><td align=center>Out 1</td><td>Right Channel</td></tr>
|
|
</table>
|
|
<h3>Functions</h3>
|
|
<p>This object has no functions to call from the Arduino sketch. It
|
|
simply streams data from the USB to its 2 output ports.</p>
|
|
<!--
|
|
<h3>Hardware</h3>
|
|
-->
|
|
<h3>Examples</h3>
|
|
<p class=exam>File > Examples > Audio > HardwareTesting > PassThroughUSB</p>
|
|
</p>
|
|
<h3>Notes</h3>
|
|
<p>Arduino's <b>Tools > USB Type</b> menu must be set to <b>Audio</b>.
|
|
</p>
|
|
<p align=center><img src="img/usbtype_audio.png"></p>
|
|
<p>USB input & output does not cause the Teensy Audio Library to
|
|
update. At least one non-USB input or output object must be
|
|
present for the entire library to update properly.</p>
|
|
<p>A known problem exists with USB audio from Macintosh computers.
|
|
An imperfect <a href="https://forum.pjrc.com/threads/34855-Distorted-audio-when-using-USB-input-on-Teensy-3-1?p=110392&viewfull=1#post110392">workaround
|
|
can be enabled by editing usb_audio.cpp</a>.
|
|
Find and uncomment "#define MACOSX_ADAPTIVE_LIMIT".</p>
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioInputUSB_F32">
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
<script type="text/x-red" data-help-name="AudioOutputUSB_F32">
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>Send stereo audio to a PC or Mac. Teensy appears as a USB
|
|
sound device.</p>
|
|
<p align=center><img src="img/usbtype_audio_out.png"></p>
|
|
</div>
|
|
<h3>Audio Connections</h3>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=top><th>Port</th><th>Purpose</th></tr>
|
|
<tr class=odd><td align=center>In 0</td><td>Left Channel</td></tr>
|
|
<tr class=odd><td align=center>In 1</td><td>Right Channel</td></tr>
|
|
</table>
|
|
<h3>Functions</h3>
|
|
<p>This object has no functions to call from the Arduino sketch. It
|
|
simply streams from it's 2 input ports to the USB.</p>
|
|
<!--
|
|
<h3>Hardware</h3>
|
|
-->
|
|
<h3>Examples</h3>
|
|
<p class=exam>File > Examples > Audio > HardwareTesting > WavFilePlayerUSB</p>
|
|
</p>
|
|
<h3>Notes</h3>
|
|
<p>Arduino's <b>Tools > USB Type</b> menu must be set to <b>Audio</b>.
|
|
</p>
|
|
<p align=center><img src="img/usbtype_audio.png"></p>
|
|
<p>USB input & output does not cause the Teensy Audio Library to
|
|
update. At least one non-USB input or output object must be
|
|
present for the entire library to update properly.</p>
|
|
<p>A known problem exists with USB audio from Macintosh computers.
|
|
An imperfect <a href="https://forum.pjrc.com/threads/34855-Distorted-audio-when-using-USB-input-on-Teensy-3-1?p=110392&viewfull=1#post110392">workaround
|
|
can be enabled by editing usb_audio.cpp</a>.
|
|
Find and uncomment "#define MACOSX_ADAPTIVE_LIMIT".</p>
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioOutputUSB_F32">
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
<script type="text/x-red" data-help-name="AudioPlayQueue_F32">
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>Play audio data provided by the Arduino sketch. This object provides
|
|
functions to allow the sketch code to push data into the audio system.</p>
|
|
</div>
|
|
<h3>Audio Connections</h3>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=top><th>Port</th><th>Purpose</th></tr>
|
|
<tr class=odd><td align=center>Out 0</td><td>Sound Output</td></tr>
|
|
</table>
|
|
<h3>Functions</h3>
|
|
<p class=func><span class=keyword>play</span>(int16);</p>
|
|
<p class=desc>not yet implemented
|
|
</p>
|
|
<p class=func><span class=keyword>play</span>(int16[], length);</p>
|
|
<p class=desc>not yet implemented
|
|
</p>
|
|
<p class=func><span class=keyword>getBuffer</span>();</p>
|
|
<p class=desc>Returns a pointer to an array of 128 int16. This buffer
|
|
is within the audio library memory pool, providing the most efficient
|
|
way to input data to the audio system. The buffer is likely to be
|
|
populated by previously used data, so the entire 128 words should be
|
|
written before calling playBuffer(). Only a single buffer should be
|
|
requested at a time. This function may return NULL if no memory is
|
|
available.
|
|
</p>
|
|
<p class=func><span class=keyword>playBuffer</span>();</p>
|
|
<p class=desc>Transmit the buffer previously obtained from getBuffer().
|
|
</p>
|
|
<h3>Examples</h3>
|
|
<p><a href="http://community.arm.com/groups/embedded/blog/2014/05/23/led-video-panel-at-maker-faire-2014" target="_blank">4320 LED Video+Sound Project</a>
|
|
</p>
|
|
<!--
|
|
<p class=exam>File > Examples > Audio >
|
|
</p>
|
|
-->
|
|
<h3>Notes</h3>
|
|
<p>TODO: many caveats....</p>
|
|
<p>
|
|
</p>
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioPlayQueue_F32">
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
<script type="text/x-red" data-help-name="AudioRecordQueue_F32">
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>Record audio data by sending to the Arduino sketch. This object allows
|
|
sketch code to receive audio packets.</p>
|
|
</div>
|
|
<h3>Audio Connections</h3>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=top><th>Port</th><th>Purpose</th></tr>
|
|
<tr class=odd><td align=center>In 0</td><td>Sound To Access</td></tr>
|
|
</table>
|
|
<h3>Functions</h3>
|
|
<p class=func><span class=keyword>begin</span>();</p>
|
|
<p class=desc>Begin capturing incoming audio to the queue. After calling
|
|
begin, readBuffer() and freeBuffer(), or clear() must be used frequently
|
|
to prevent the queue from filling up.
|
|
</p>
|
|
<p class=func><span class=keyword>available</span>();</p>
|
|
<p class=desc>Returns the number of audio packets available to read.
|
|
</p>
|
|
<p class=func><span class=keyword>readBuffer</span>();</p>
|
|
<p class=desc>Read a single audio packet. A pointer to a 128 sample
|
|
array of 16 bit integers is returned. NULL is returned if no packets
|
|
are available.
|
|
</p>
|
|
<p class=func><span class=keyword>freeBuffer</span>();</p>
|
|
<p class=desc>Release the memory from the previously read packet returned
|
|
from readBuffer(). Only a single packet at a time may be read, and
|
|
each packet must be freed with this function, to return the memory to
|
|
the audio library.
|
|
</p>
|
|
<p class=func><span class=keyword>clear</span>();</p>
|
|
<p class=desc>Discard all audio held in the queue.
|
|
</p>
|
|
<p class=func><span class=keyword>end</span>();</p>
|
|
<p class=desc>Stop capturing incoming audio into the queue. Data already
|
|
captured remains in the queue and may be read with readBuffer().
|
|
</p>
|
|
<h3>Examples</h3>
|
|
<p class=exam>File > Examples > Audio > Recorder
|
|
</p>
|
|
<h3>Notes</h3>
|
|
<p>
|
|
Up to 52 packets may be queued by this object, which allows approximately
|
|
150 ms of audio to be held in the queue, to allow time for the Arduino
|
|
sketch to write data to media or do other high-latency tasks.
|
|
|
|
The actual packets are taken
|
|
from the pool created by AudioMemory().
|
|
</p>
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioRecordQueue_F32">
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
<script type="text/x-red" data-help-name="AudioSynthNoisePink_F32">
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>Create pink noise, using Stefan Stenzel's "New Shade Of Pink" algorithm.
|
|
</p>
|
|
<!--
|
|
<p align=center><img src="img/whitenoise.png"></p>
|
|
-->
|
|
</div>
|
|
<h3>Audio Connections</h3>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=top><th>Port</th><th>Purpose</th></tr>
|
|
<tr class=odd><td align=center>Out 0</td><td>Pink Noise</td></tr>
|
|
</table>
|
|
<h3>Functions</h3>
|
|
<p class=func><span class=keyword>amplitude</span>(level);</p>
|
|
<p class=desc>Set the output peak level, from 0 (off) to 1.0.
|
|
The default is off. Noise is generated only after setting
|
|
to a non-zero level.
|
|
</p>
|
|
<h3>Examples</h3>
|
|
<p class=exam>File > Examples > Audio > MemoryAndCpuUsage
|
|
</p>
|
|
<h3>Notes</h3>
|
|
<p>Setting the amplitude to zero causes this object to stop using
|
|
CPU time. CPU usage is approx 3% on Teensy 3.1.
|
|
</p>
|
|
<p>Stefan Stenzel's
|
|
<a href="http://stenzel.waldorfmusic.de/post/pink/" target="_blank">New Shade Of Pink</a>
|
|
algorithm. Stefan's terms of use are "Use for any purpose. If used
|
|
in a commercial product, you should give me one."
|
|
</p>
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioSynthNoisePink_F32">
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
<script type="text/x-red" data-help-name="AudioSynthWaveformSine_F32">
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>Create a sine wave signal</p>
|
|
</div>
|
|
<h3>Audio Connections</h3>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=top><th>Port</th><th>Purpose</th></tr>
|
|
<tr class=odd><td align=center>Out 0</td><td>Sine Wave Output</td></tr>
|
|
</table>
|
|
<h3>Functions</h3>
|
|
<p class=func><span class=keyword>amplitude</span>(level);</p>
|
|
<p class=desc> <p class=desc>The amplitude, level, is the peak, as in
|
|
zero-to-peak. This produces an output ranging from -a to +a.
|
|
</p>
|
|
Negative values do a 180-degree phase reversal.
|
|
</p>
|
|
<p class=func><span class=keyword>frequency</span>(freq);</p>
|
|
<p class=desc>Set the frequency, from 0.0 to half of the sampling
|
|
frequency. This is a floating point number and thus not limited to integers.
|
|
Values such as 123.456 will be set properly.
|
|
</p>
|
|
<p class=func><span class=keyword>phase</span>(angle);</p>
|
|
<p class=desc>
|
|
Cause the generated waveform to jump to a specific point within
|
|
its cycle. Angle is from 0.0 to 360.0 degrees. When multiple objects
|
|
are configured,
|
|
<a href="http://www.pjrc.com/teensy/td_libs_AudioProcessorUsage.html" target="_blank">AudioNoInterrupts()</a>
|
|
should be used to guarantee all new settings take effect together.
|
|
</p>
|
|
<p class=func><span class=keyword>setSampleRate_Hz</span>(float &fs_Hz);</p>
|
|
<p class=desc>Sets sample rate for this class only. Default 44.1 kHz.</p>
|
|
<p class=func><span class=keyword>begin</span>(void);</p>
|
|
<p class=desc>Defaults to sine running. But, along with the end() function
|
|
can be used to turn the wave on and off.
|
|
</p>
|
|
<p class=func><span class=keyword>end</span>(void);</p>
|
|
<p class=func><span class=keyword>pureSpectrum</span>(bool _setPure);</p>
|
|
<p class=desc>The function pureSpectrum(true) enables two stages of
|
|
biquad filtering putting the harmonics generally below -135 dBc. This filter
|
|
tracks the frequency() entry, and is available above a few hundred Hz,
|
|
depending on the sample rate.
|
|
</p>
|
|
<h3>Examples</h3>
|
|
<p class=exam>File > Examples > ReceiverPart1
|
|
</p>
|
|
<p class=exam>File > Examples > AudioTestPeakRMS
|
|
</p>
|
|
<p class=exam>File > Examples > LowpassFilter_FD_OA
|
|
</p>
|
|
<h3>Notes</h3>
|
|
<p></p>
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioSynthWaveformSine_F32">
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
|
|
|
|
<script type="text/x-red" data-help-name="AudioSynthSineCosine_F32">
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>Creates both a sine wave and cosine signals with 90 degree
|
|
phase difference. 90 degree difference can be adjusted. Both have the
|
|
same adjustable amplitude. </p>
|
|
</div>
|
|
<h3>Audio Connections</h3>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=top><th>Port</th><th>Purpose</th></tr>
|
|
<tr class=odd><td align=center>Out 0</td><td>Sine Wave Output</td></tr>
|
|
<tr class=odd><td align=center>Out 0</td><td>Cosine Wave Output</td></tr>
|
|
</table>
|
|
<h3>Functions</h3>
|
|
<p class=func><span class=keyword>amplitude</span>(<strong>float </strong>level);</p>
|
|
<p class=desc>The amplitude, a, is the peak, as in zero-to-peak. This produces outputs
|
|
ranging from -a to +a. Both outputs are the same amplitude.
|
|
</p>
|
|
|
|
<p class=func><span class=keyword>frequency</span>(<strong>float </strong>freq);</p>
|
|
<p class=desc>Set the frequency of the sine and cosine waveforms, in Hz. Both
|
|
are the same.</p>
|
|
|
|
<p class=func><span class=keyword>phase</span>(<strong>float </strong>angle);</p>
|
|
<p class=desc>
|
|
Cause the generated waveform to jump to a specific point within
|
|
its cycle. Angle is from 0 to 360 degrees. When multiple objects
|
|
are configured,
|
|
<a href="http://www.pjrc.com/teensy/td_libs_AudioProcessorUsage.html" target="_blank">AudioNoInterrupts()</a>
|
|
should be used to guarantee all new settings take effect together.</p>
|
|
|
|
|
|
<p class=func><span class=keyword>phase_r</span>(<strong>float </strong> phr)
|
|
<p class=desc>Externally, phase comes in the range (0,2*M_PI) keeping with C math functions
|
|
Internally, the full circle is represented as (0.0, 512.0). This is
|
|
convenient for finding the entry to the sine table.</p>
|
|
|
|
|
|
|
|
<p class=func><span class=keyword>phaseS_C_r</span>(<strong>float </strong> phsc)
|
|
<p class=desc>phaseS_C_r is the number of radians that the cosine output leads the
|
|
sine output. The default is M_PI_2 = pi/2 = 1.57079633 radians,
|
|
corresponding to 90.00 degrees cosine leading sine.</p>
|
|
|
|
|
|
<p class=func><span class=keyword>void simple</span>(<strong>bool </strong> s)
|
|
<p class=desc>Speed up calculations by setting phaseS_C=90deg, amplitude=1
|
|
Note, s=true will override any setting of phaseS_C_r or amplitude.</p>
|
|
|
|
<p class=func><span class=keyword>pureSpectrum</span>(bool _setPure);</p>
|
|
<p class=desc>The function pureSpectrum(true) enables two stages of
|
|
biquad filtering on each output, putting the harmonics generally below -135 dBc.
|
|
This filter tracks the frequency() entry, and is available above a few hundred Hz,
|
|
depending on the sample rate.
|
|
</p>
|
|
|
|
<p class=func><span class=keyword>setSampleRate_Hz</span>(<strong>float </strong> fs_Hz)
|
|
<p class=desc>Sets sample rate for this class only. Default 44.1 kHz.</p>
|
|
|
|
|
|
<p class=func><span class=keyword>setBlockLength</span>(<strong>uint16_t </strong> bl)
|
|
<p class=desc>Sets block length for this class only. Default 128.</p>
|
|
|
|
|
|
|
|
</p>
|
|
<h3>Examples</h3>
|
|
<p class=exam>File > Examples > Audio > MemoryAndCpuUsage
|
|
</p>
|
|
<p class=exam>File > Examples > Audio > Analysis > ---
|
|
</p>
|
|
<p class=exam>File > Examples > Audio > Analysis > ---
|
|
</p>
|
|
<h3>Notes</h3>
|
|
<p></p>
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioSynthSineCosine_F32">
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
|
|
|
|
<script type="text/x-red" data-help-name="AudioSynthWaveform_F32">
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>Create a waveform: sine, sawtooth, square, triangle, pulse or arbitrary.</p>
|
|
</div>
|
|
<h3>Audio Connections</h3>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=top><th>Port</th><th>Purpose</th></tr>
|
|
<tr class=odd><td align=center>Out 0</td><td>Waveform Output</td></tr>
|
|
</table>
|
|
<h3>Functions</h3>
|
|
<p class=func><span class=keyword>begin</span>(waveform);</p>
|
|
<p class=desc>Configure the waveform type to create.
|
|
</p>
|
|
<p class=func><span class=keyword>begin</span>(level, frequency, waveform);</p>
|
|
<p class=desc>Output a waveform, and set the amplitude and frequency.
|
|
</p>
|
|
<p class=func><span class=keyword>frequency</span>(freq);</p>
|
|
<p class=desc>Change the frequency.
|
|
</p>
|
|
<p class=func><span class=keyword>amplitude</span>(level);</p>
|
|
<p class=desc>Change the amplitude. Set to 0 to turn the signal off.
|
|
</p>
|
|
<p class=func><span class=keyword>phase</span>(angle);</p>
|
|
<p class=desc>
|
|
Cause the generated waveform to jump to a specific point within
|
|
its cycle. Angle is from 0 to 360 degrees. When multiple objects
|
|
are configured,
|
|
<a href="http://www.pjrc.com/teensy/td_libs_AudioProcessorUsage.html" target="_blank">AudioNoInterrupts()</a>
|
|
should be used to guarantee all new settings take effect together.
|
|
</p>
|
|
<p class=func><span class=keyword>pulseWidth</span>(amount);</p>
|
|
<p class=desc>Change the width (duty cycle) of the pulse.</p>
|
|
<p class=func><span class=keyword>arbitraryWaveform</span>(array, maxFreq);</p>
|
|
<p class=desc>
|
|
Configure the waveform to be used with WAVEFORM_ARBITRARY. Array
|
|
must be an array of 256 samples. Currently, the data is used
|
|
without any filtering, which can cause aliasing with frequencies
|
|
above 172 Hz. For higher frequency output, you must bandwidth
|
|
limit your waveform data. Someday, "maxFreq" will be used to
|
|
do this automatically.
|
|
</p>
|
|
<h3>Examples</h3>
|
|
<p class=exam>File > Examples > Audio > Synthesis > PlaySynthMusic
|
|
</p>
|
|
<p class=exam>File > Examples > Audio > Synthesis > pulseWidth
|
|
</p>
|
|
<p class=exam>File > Examples > Audio > HardwareTesting > WM8731MikroSine
|
|
</p>
|
|
<h3>Notes</h3>
|
|
<p>Supported Waveforms:<br>
|
|
<ul>
|
|
<li><span class=literal>WAVEFORM_SINE</span></li>
|
|
<li><span class=literal>WAVEFORM_SAWTOOTH</span></li>
|
|
<li><span class=literal>WAVEFORM_SAWTOOTH_REVERSE</span></li>
|
|
<li><span class=literal>WAVEFORM_SQUARE</span></li>
|
|
<li><span class=literal>WAVEFORM_TRIANGLE</span></li>
|
|
<li><span class=literal>WAVEFORM_ARBITRARY</span></li>
|
|
<li><span class=literal>WAVEFORM_PULSE</span></li>
|
|
<li><span class=literal>WAVEFORM_SAMPLE_HOLD</span></li>
|
|
</ul>
|
|
</p>
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioSynthWaveform_F32">
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
<div>
|
|
<script type="text/x-red" data-help-name="AudioSynthGaussian_F32">
|
|
<!-- ============ AudioSynthGaussian_F32 ========= -->
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>Generates Gaussian White Noise of known power.</p>
|
|
</div>
|
|
<h3>Boards Supported</h3>
|
|
<ul>
|
|
<li>Teensy 3.2
|
|
<li>Teensy 3.5
|
|
<li>Teensy 3.6
|
|
<li>Teensy 4.0
|
|
<li>Teensy 4.1
|
|
</ul>
|
|
<h3>Audio Connections</h3>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=top><th>Port</th><th>Purpose</th></tr>
|
|
<tr class=odd><td align=center>Out 0</td><td>Output Noise Signal</td></tr>
|
|
</table>
|
|
|
|
<h3>Functions</h3>
|
|
<p class=func><span class=keyword>amplitude</span>(<strong>float</strong> sd);</p>
|
|
<p class=desc>Sets the amplitude of the noise. This is the 1-sima or standard
|
|
deviation level. Since this is Gaussian distributed noise, the same number is also
|
|
the RMS level (square root of power).
|
|
|
|
<h3>Examples</h3>
|
|
<p class=exam>File > Examples > OpenAudio_ArduinoLibrary > TestNoiseBlanker1
|
|
</p>
|
|
|
|
<h3>Notes</h3>
|
|
<p>This noise source is a Gaussian distribution with mean of
|
|
zero and a standard deviation specified by amplitude(). Default amlitude
|
|
is 0.0 (off). Individual samples are fully uncorrelated, meaning that
|
|
the spectrum of the noise is flat (White Noise).</p>
|
|
|
|
<p>Time requirements for generating a block of 128, Teensy 3.6,
|
|
is 121 microseconds and for the Teensy 4.0 is 36 microseconds.</p>
|
|
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioSynthGaussian_F32">>
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
|
|
|
|
|
|
<script type="text/x-red" data-help-name="AudioSynthNoiseWhite_F32">
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>Create white noise.
|
|
</p>
|
|
<p align=center><img src="img/whitenoise.png"></p>
|
|
</div>
|
|
<h3>Audio Connections</h3>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=top><th>Port</th><th>Purpose</th></tr>
|
|
<tr class=odd><td align=center>Out 0</td><td>White Noise</td></tr>
|
|
</table>
|
|
<h3>Functions</h3>
|
|
<p class=func><span class=keyword>amplitude</span>(level);</p>
|
|
<p class=desc>Set the output peak level, from 0 (off) to 1.0.
|
|
The default is off. Noise is generated only after setting
|
|
to a non-zero level.
|
|
</p>
|
|
<h3>Examples</h3>
|
|
<p class=exam>File > Examples > Audio >
|
|
</p>
|
|
<h3>Notes</h3>
|
|
<p>Setting the amplitude to zero causes this object to stop using
|
|
CPU time to generate random numbers.
|
|
</p>
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioSynthNoiseWhite_F32">
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
|
|
|
|
<div>
|
|
<script type="text/x-red" data-help-name="RadioFMDetector_F32">
|
|
<!-- ============ RadioFMDetector_F32 ========= -->
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>An FM Detector sutable for work at an low frequency such
|
|
as 15 kHz. An output low-pass filter is included. A squelch
|
|
allows for silencing noise when signals are not present.
|
|
For deviations in the 10 kHz range or less. Not for wideband
|
|
broadcast FM.
|
|
</p>
|
|
</div>
|
|
<h3>Boards Supported</h3>
|
|
<ul>
|
|
<li>Teensy 3.2
|
|
<li>Teensy 3.5
|
|
<li>Teensy 3.6
|
|
<li>Teensy 4.0
|
|
<li>Teensy 4.1
|
|
</ul>
|
|
<h3>Audio Connections</h3>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=top><th>Port</th><th>Purpose</th></tr>
|
|
<tr class=odd><td align=center>In 0</td><td>Input Signal</td></tr>
|
|
<tr class=odd><td align=center>Out 0</td><td>De-modulated Output Signal</td></tr>
|
|
<tr class=odd><td align=center>Out 1</td><td>De-modulated Output Signal, squelched</td></tr>
|
|
</table>
|
|
|
|
<h3>Functions</h3>
|
|
<p class=func><span class=keyword>frequency</span>(<strong>float</strong> fCenter);</p>
|
|
<p class=desc></p>
|
|
<p>Sets the center frequency, in Hz.</p>
|
|
|
|
<p class=func><span class=keyword>filterOut</span>(<strong>float</strong> *firCoeffs, <strong>uint</strong> nFIR, <strong>float</strong> Kdem);</p>
|
|
<p>This sets output filtering where:
|
|
<pre class="desc">
|
|
float32_t* firCoeffs pointer to array of coefficients
|
|
uint nFIR is the number of coefficients
|
|
float32_t Kdem is the de-emphasis frequency factor
|
|
Kdem = 1/(0.5+(tau*fsample))
|
|
tau is the de-emphasis time constant,
|
|
typically 0.0005 second and fsample
|
|
the sample frequency, typically 44117.
|
|
</pre>
|
|
<p class=func><span class=keyword>filterIQ</span>(<strong>float</strong> *fir_IQ_Coeffs, <strong>uint</strong> nFIR_IQ);</p>
|
|
<p>This sets the detector output filtering where:
|
|
<pre class="desc">
|
|
float32_t* fir_IQ_Coeffs is an array of coefficients
|
|
uint nFIR_IQ is the number of coefficients, max 60
|
|
</pre>
|
|
|
|
<p class=func><span class=keyword>setSquelchThreshold</span>(<strong>float</strong> sqThresh);</p>
|
|
<p class=desc></p>
|
|
<p>Sets the squelch threshold ranging 0.0 to 1.0 where
|
|
0.0 always lets audio through.</p>
|
|
|
|
<p class=func><span class=keyword>setSquelchDecay </span>(<strong>float</strong> sqDecay);</p>
|
|
<p class=desc></p>
|
|
<p>Sets the decay rate of the output of the squelch detector. This produces the "squelch tail."
|
|
The range is 0.9 (no real tail) to 0.9999 (very slow decay and a long squelch tail.)
|
|
The default value is 0.99.</p>
|
|
|
|
<p class=func><span class=keyword>getSquelchLevel</span>();</p>
|
|
<p class=desc></p>
|
|
<p>Returns the current measured squelch level as a <strong>float. </strong>
|
|
Higher levels, towards 1.0, are no signal. A low return value indicates presence of
|
|
a signal.</p>
|
|
|
|
<p class=func><span class=keyword>setSquelchFilter</span>(<strong>float* </strong>Coefficients);</p>
|
|
<p class=desc></p>
|
|
<p>This allows changing the 2-section (4-pole) bandpass BiQuad filter used before the
|
|
squelch detector. This passes high-frequency noise and attenuates lower frequency voice. The
|
|
parameter "Coefficients" points to an array of 10 floating point numbers. A Coefficient value
|
|
of NULL restores the default 3 to 5 kHz filter. See the example ReceiverFM.ino for using this.</p>
|
|
|
|
<p class=func><span class=keyword>returnInitializeFMError</span>();</p>
|
|
<p>This returns the initialization errors, for debug use.
|
|
<pre class="desc">
|
|
B0001 (value 1) error in IQ FIR Coefficients or qty
|
|
B0010 (value 2) error in Output FIR Coefficients or qty
|
|
B0100 (value 4) error in de-emphasis constant
|
|
B1000 (value 8) errors center frequency to high
|
|
</pre>
|
|
|
|
<h3>Examples</h3>
|
|
<p class=exam>File > Examples > OpenAudio_ArduinoLibrary > ReceiverFM
|
|
</p>
|
|
|
|
<p class=exam>File > Examples > OpenAudio_ArduinoLibrary > ReceiverPart2
|
|
</p>
|
|
|
|
<h3>Notes</h3>
|
|
<p>This consists of a single input at some frequency, such as 10 to 20 kHz and
|
|
an output, such as 0 to 5 kHz. The output level is linearly dependent on the
|
|
frequency of the input sine wave frequency, i.e., an it is an FM detector.
|
|
The input needs to be band limited below the lower frequency side of the
|
|
input, typically 10 kHz. This is not part of this block.</p>
|
|
|
|
<p>This uses 430 microseconds for an 128 point update with Teensy 3.6. With the
|
|
Teensy 4.x, 97 microseconds is used.</p>
|
|
|
|
<P>The output can be FIR filtered using default parameters,
|
|
or using coefficients from an array. A separate single pole de-emphasis filer
|
|
is included that again can be programmed.</P>
|
|
|
|
<p><strong>Output:</strong>Float, sensitivity is 2*pi*(f - fCenter)*sample_rate_Hz
|
|
For 44117Hz sample rate, this is 0.000142421 per Hz.</p>
|
|
|
|
<p><strong>Accuracy</strong>The function used is precise. However, the approximations, such
|
|
as fastAtan2, slightly limit the accuracy. A 200 point sample of a
|
|
14 kHz input had an average error of 0.03 Hz
|
|
and a standard deviation of 0.81 Hz.</p>
|
|
|
|
<p>The RadioFMDetector_F32.h file has much more information including details
|
|
of the arc-tangent method of computation.</p>
|
|
|
|
</script>
|
|
<script type="text/x-red" data-template-name="RadioFMDetector_F32">>
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
<div>
|
|
<script type="text/x-red" data-help-name="radioModulatedGenerator_F32">
|
|
<!-- ============ radioModulatedGenerator_F32 ========= -->
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>A modulator to apply AM, PM or FM to a carrier sine wave. Outputs
|
|
can be single waveform, or a pair of waveforms of I and Q form for
|
|
external quadrature up conversion. The latter includes corrections for
|
|
amplitude and phase errors.
|
|
</p>
|
|
</div>
|
|
<h3>Boards Supported</h3>
|
|
<ul>
|
|
<li>Teensy 3.2
|
|
<li>Teensy 3.5
|
|
<li>Teensy 3.6
|
|
<li>Teensy 4.0
|
|
<li>Teensy 4.1
|
|
</ul>
|
|
<h3>Audio Connections</h3>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=top><th>Port</th><th>Purpose</th></tr>
|
|
<tr class=odd><td align=center>In 0</td><td>AM Modulating Signal</td></tr>
|
|
<tr class=odd><td align=center>In 1</td><td>PM or FM Modulating Signal</td></tr>
|
|
<tr class=odd><td align=center>Out 0</td><td>Single or I Output Signal</td></tr>
|
|
<tr class=odd><td align=center>Out 1</td><td>Q Output Signal</td></tr>
|
|
</table>
|
|
|
|
<h3>Functions</h3>
|
|
<p class=func><span class=keyword>frequency</span>(<strong>float</strong> fCarrier);</p>
|
|
<p class=desc></p>
|
|
<p>Sets the center carrier frequency, in Hz.</p>
|
|
|
|
<p class=func><span class=keyword>amplitude</span>(<strong>float</strong> ampl);</p>
|
|
<p class=desc></p>
|
|
<p>Sets the carrier amplitude level. For AM, the peak output will be greater than this
|
|
level, reaching twice level at 100% modulation.</p>
|
|
|
|
<p class=func><span class=keyword>phase_r</span>(<strong>float</strong> ph);</p>
|
|
<p class=desc></p>
|
|
<p>Sets the carrier starting phase in radians, 0 to 2*pi. Used with multiple
|
|
modulated generators to initialize the relative phase values. Stop interrupts before setting.</p>
|
|
|
|
<p class=func><span class=keyword>doModulation_AM_PM_FM</span>(<strong>bool</strong> doAM, <strong>bool</strong> doPM, <strong>bool</strong> doFM, <strong>bool</strong> bothIQ);</p>
|
|
<p class=desc></p>
|
|
<p>Selects modulation format. PM and FM cannot be used together. Otherwise the selection is flexible.
|
|
For instance, AM and PM can be used together for QAM modulation. bothIQ selects whether there is a single
|
|
modulated output (false) or a pair of quadrature outputs to drive external hardware (true).</p>
|
|
|
|
<p class=func><span class=keyword>phaseQ_I_r(</span>(<strong>float</strong> ph_IQ);</p>
|
|
<p class=desc></p>
|
|
<p>Used with bothIQ==true to set the relative phase of the outputs. The default
|
|
is PI/2 corresponding to 90 degrees. This can correct for phase errors in external hardware.</p>
|
|
|
|
<p class=func><span class=keyword>amplitudeQI(</span>(<strong>float</strong> ampl_IQ);</p>
|
|
<p class=desc></p>
|
|
<p>Used with bothIQ==true to set the relative amplitude of the outputs. The default
|
|
is 1.0. This can correct for amplitude errors in external hardware.</p>
|
|
|
|
<h3>Examples</h3>
|
|
<p class=exam>File > Examples > OpenAudio_ArduinoLibrary > AM_PM_FM.ino
|
|
</p>
|
|
|
|
<h3>Notes</h3><p>For AM, 100% AM modulation corresponds
|
|
an input of -1.0 to 1.0, regardless of the
|
|
carrier amplitude setting. Overmodulation (more that 100%) results in peak
|
|
increases beyond twice amplitude, but full abrupt clipping at the
|
|
bottom zero point. Clipping on the top would be in an external block,
|
|
if desired.</p>
|
|
|
|
<p> <pre class="desc">
|
|
Times: T3.6 update() block of 128 is about 53 microseconds, AM Single output.
|
|
T4.x update() block of 128 is about 20 microseconds AM Single output
|
|
T4.x update() block of 128 is about 35 microseconds AM I + Q outputs
|
|
For T4.x, FM is 1 or 2 microseconds faster than AM.
|
|
</pre>
|
|
|
|
</script>
|
|
<script type="text/x-red" data-template-name="radioModulatedGenerator_F32">>
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
<div>
|
|
<script type="text/x-red" data-help-name="RadioIQMixer_F32">
|
|
<!-- ============ RadioIQMixer_F32 ========= -->
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>This quadrature mixer block for both transmit and receive.
|
|
A basic building block is a pair of mixers with the
|
|
LO going to the mixers at the same frequency, but differing in phase
|
|
by 90 degrees (programmable). The LO's are included
|
|
in the block, but there are no post-mixing filters. </p>
|
|
</div>
|
|
<h3>Boards Supported</h3>
|
|
<ul>
|
|
<li>Teensy 3.5
|
|
<li>Teensy 3.6
|
|
<li>Teensy 4.0
|
|
<li>Teensy 4.1
|
|
</ul>
|
|
<h3>Audio Connections</h3>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=top><th>Port</th><th>Purpose</th></tr>
|
|
<tr class=odd><td align=center>In 0</td><td>Input I Signal</td></tr>
|
|
<tr class=odd><td align=center>In 1</td><td>Input Q Signal</td></tr>
|
|
<tr class=odd><td align=center>Out 0</td><td>I In-Phase Out</td></tr>
|
|
<tr class=odd><td align=center>Out 1</td><td>Q Quadrature Output</td></tr>
|
|
</table>
|
|
|
|
<h3>Functions</h3>
|
|
|
|
<p class=func><span class=keyword>frequency</span>(<strong>float</strong> fr);</p>
|
|
<p class=desc></p>
|
|
<p>Sets Mixer LO frequency in Hz</p>
|
|
|
|
<p class=func><span class=keyword>iqmPhaseS</span>(<strong>float</strong> ps);</p>
|
|
<p class=desc>This phase comes in the range (0, 2PI) keeping with C math functions.
|
|
This function allows multiple mixers to be phase coordinated (stop
|
|
interrupts when setting). </p>
|
|
|
|
<p class=func><span class=keyword>phaseS_C_r</span>(<strong>float</strong> pc);</p>
|
|
<p class=desc> Sets the number of radians that the cosine LO leads the
|
|
sine LO. The default is PI/2 radians. This is used to correct hardware phase unbalance.</p>
|
|
|
|
<p class=func><span class=keyword>amplitudeC</span>(<strong>float</strong> g);</p>
|
|
<p class=desc> Sets the gain, g, for the I channel.
|
|
The Q channel is always 1.0. This is used to correct hardware amplitude unbalance.</p>
|
|
|
|
|
|
<p class=func><span class=keyword>void useTwoChannel</span>(<strong>bool</strong> twoCh);</p>
|
|
<p class=desc>
|
|
Channel 0 (left) is the in-phase input I for twoCh true of false. Channel 1 (right) is Q for
|
|
complex 2-channel input (twoCh==true) and not used for twoChannel==false. Caution, never
|
|
use twoCh=false with two inputs.
|
|
</p>
|
|
|
|
<p class=func><span class=keyword>useSimple</span>(<strong>bool</strong> simple);</p>
|
|
<p class=desc>Faster if true, but no phase/amplitude adjustment.</p>
|
|
|
|
<h3>Examples</h3>
|
|
<p class=exam>File > Examples > OpenAudio_ArduinoLibrary > ReceiverPart1
|
|
</p>
|
|
|
|
<p class=exam>File > Examples > OpenAudio_ArduinoLibrary > ReceiverPart2
|
|
</p>
|
|
|
|
<p class=exam>File > Examples > OpenAudio_ArduinoLibrary > ReceiverSSB
|
|
</p>
|
|
|
|
<h3>Notes</h3>
|
|
<p>There is provision for varying
|
|
the phase between the sine and cosine oscillators. The relative gain in the
|
|
I and Q channels is also programmable. This allows for flaws in the
|
|
response of real hardware.</p>
|
|
|
|
<P>The output levels are 0.5 times the input level </P>
|
|
|
|
<p>Time: T3.6, For an update of a 128 sample block, doSimple=1, 46 microseconds;
|
|
T4.0, For an update of a 128 sample block, doSimple=1, 20 microseconds</p>
|
|
|
|
</script>
|
|
<script type="text/x-red" data-template-name="RadioIQMixer_F32">>
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
|
|
<div>
|
|
<script type="text/x-red" data-help-name="radioNoiseBlanker_F32">
|
|
<!-- ============ radioNoiseBlanker_F32 ========= -->
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>Looks for wideband impulse noise and cuts off the flow of audio when an impulse
|
|
is detected. A delay in the audio allows anticipation of impulse noise.
|
|
The threshold of impulse detection is adjusted by the averaged
|
|
audio level. Parameters are programmable. Single Path or I-Q.</p>
|
|
</div>
|
|
<h3>Boards Supported</h3>
|
|
<ul>Teensy 3.5
|
|
<li>Teensy 3.6
|
|
<li>Teensy 4.0
|
|
<li>Teensy 4.1
|
|
</ul>
|
|
<h3>Audio Connections</h3>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=top><th>Port</th><th>Purpose</th></tr>
|
|
<tr class=odd><td align=center>In 0</td><td>Input Signal 0</td></tr>
|
|
<tr class=odd><td align=center>In 1</td><td>Input Signal 1 (optional)</td></tr>
|
|
<tr class=odd><td align=center>Out 0</td><td>Output Signal 0 with Blanking</td></tr>
|
|
<tr class=odd><td align=center>Out 1</td><td>Output Sig 1 for Input 1</td></tr>
|
|
</table>
|
|
|
|
<h3>Functions</h3>
|
|
<p class=func><span class=keyword>setNoiseBlanker</span>(<strong>float</strong> threshold, <strong>uint16_t</strong> nAnticipation, <strong>uint16_t</strong> nDecay);</p>
|
|
<p class=desc>Sets the three parameters of the noise blanker.
|
|
The variable threshold adjusts the minimum level considered to be an impulse.
|
|
Anticipation is the number of samples of delay for the signal (1, 125)
|
|
and nDecay is the number of samples that are included in the blanked
|
|
period after the impulse has dropped below the threshold (1, 10).</p>
|
|
|
|
<p class=func><span class=keyword>enable</span>(<strong>bool</strong> e);</p>
|
|
<p class=desc>If e==true the noise blanker is on and if e==false it
|
|
is off (default). This is the on/off switch.</p>
|
|
|
|
<p class=func><span class=keyword>useTwoChannel</span>(<strong>bool</strong> twoCh);</p>
|
|
<p class=desc>If twoCh==true, input 1 becomes active and is blanked at the same
|
|
time as the signal of input 0. This is for I-Q receivers.</p>
|
|
}
|
|
|
|
<h3>Examples</h3>
|
|
<p class=exam>File > Examples > OpenAudio_ArduinoLibrary > TestNoiseBlanker1
|
|
</p>
|
|
|
|
<h3>Notes</h3>
|
|
<p>An intermediate frequency (I-F signal) comes in with occassional
|
|
noise pulses. This block watches for
|
|
the pulses and turns off the I-F while the pulse exists. In order to
|
|
be as smart as possible this looks ahead by a number of samples, nAnticipation.
|
|
Likewise, the I-F is left off for nDecay samples after the pulse ends.
|
|
Various methods could be to be used to "turn off" the I-F,
|
|
including replacement with waveforms.
|
|
As of this initial write, zeros are used in the waveform. </p>
|
|
|
|
<p>A threshold can be adjusted via setNoiseBlanker(). This is compared with the
|
|
average rectified voltage being received. If this is too small, like 1.5
|
|
or 2.0, we will be loosing good signals by blanking. If we set it too high, like 20.0,
|
|
we will not blank noise pulses. Experiments will find a good setting.
|
|
With a sine wave input and no impulse noise, the average rectified signal
|
|
should be about 0.637. To catch the top of that requires a threshold of
|
|
1/0.637 = 1.57. That would seem to be a very minimal threshold setting.</p>
|
|
|
|
</script>
|
|
<script type="text/x-red" data-template-name="radioNoiseBlanker_F32">>
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
|
|
<div>
|
|
<script type="text/x-red" data-help-name="AudioAnalyzeFFT1024_F32">
|
|
<!-- ============ AudioAnalyzeFFT1024_F32 ========= -->
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>Does real input FFT of 1024 points. Output is magnitude
|
|
only in formats of RMS (same as I16 version,
|
|
power or dBFS (full scale). Output can be bin by bin or by a pointer to
|
|
the full output array. Multiple windowing options. Uses half-lenght FFT</p>
|
|
</div>
|
|
<p>March 2021: Streamlined processing and memory useage by going to half-lenght FFT.</p>
|
|
<h3>Boards Supported</h3>
|
|
<ul>Teensy 3.5
|
|
<li>Teensy 3.6
|
|
<li>Teensy 4.0
|
|
<li>Teensy 4.1
|
|
</ul>
|
|
<h3>Audio Connections</h3>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=top><th>Port</th><th>Purpose</th></tr>
|
|
<tr class=odd><td align=center>In 0</td><td>Input Signal</td></tr>
|
|
</table>
|
|
|
|
<h3>Functions</h3>
|
|
<p class=func><span class=keyword>available</span>();</p>
|
|
<p class=desc>returns <strong>bool</strong> true if the FFT is complete,
|
|
otherwise returns false.</p>
|
|
|
|
<p class=func><span class=keyword>read</span>(<strong>int</strong> nBin);</p>
|
|
<p class=desc>Returns the output level for the specified nBin (0, 1023).
|
|
Bin 0 is DC and the bins are spaced at the sampling frequency divided
|
|
by 1024 Hz.</p>
|
|
|
|
<p class=func><span class=keyword>read</span>(<strong>int</strong> nBinFirst, <strong>int</strong> nBinLast);</p>
|
|
<p class=desc>Returns the power sum for the specified range of bins
|
|
numbered (0, 1023). This has the effect of creating a new bin with greater
|
|
width.</p>
|
|
|
|
<p class=func><span class=keyword>windowFunction</span>(<strong>int</strong> win);</p>
|
|
<p class=desc>Sets the windowing function. Instead of calling by number,
|
|
these can be called by the following defined names: </p>
|
|
<pre class="desc">
|
|
AudioWindowNone
|
|
AudioWindowHanning1024 (default)
|
|
AudioWindowBlackmanHarris1024
|
|
</pre>
|
|
|
|
<p class=func><span class=keyword>windowFunction</span>(AudioWindowKaiser1024, <strong>float</strong> kdb);</p>
|
|
<p class=desc>Sets the Kaiser window with the first sidelobe kdb
|
|
below the peak. The sidelobes continue to drop going away from a sine-wave
|
|
carrier. This is a very flexible and useful windowing function. </p>
|
|
|
|
<p class=func><span class=keyword>putWindow</span>(<strong>float</strong> *pWin);</p>
|
|
<p class=desc>Activates an INO provided window of 256 float numbers. This replaces
|
|
any window from WindowFunction(). pWin is a pointer to the window array.
|
|
|
|
<p class=func><span class=keyword>setNAverage</span>(<strong>int</strong> nAverage);</p>
|
|
<p class=desc>Sets the number of output powers that arre averaged for a single data set.
|
|
This is "non-coherent integration," or averaging. nAverage must be at least
|
|
1 with no reasonable upper limit.</p>
|
|
|
|
<p class=func><span class=keyword>getData</span>();</p>
|
|
<p class=desc>Returns a pointer to an array of 512 float outputs. This
|
|
can save 511 calls to read(). The data remains for about 10 milliseconds.</p>
|
|
|
|
<p class=func><span class=keyword>getWindow</span>();</p>
|
|
<p class=desc>Returns a pointer to an array of 1024 float points
|
|
that is the windowing function in use.</p>
|
|
|
|
<p class=func><span class=keyword>setOutputType</span>(<strong>int</strong> nType);</p>
|
|
<p class=desc>Selects the output form:</p>
|
|
<pre class="desc">
|
|
FFT_RMS 0 (default)
|
|
FFT_POWER 1
|
|
FFT_DBFS 2
|
|
</pre>
|
|
|
|
<h3>Examples</h3>
|
|
<p class=exam>File > Examples > OpenAudio_ArduinoLibrary > TestFFT1024
|
|
</p>
|
|
|
|
<h3>Notes</h3>
|
|
<p><strong>Scaling - </strong>
|
|
Full scale for floating point DSP is a nebulous concept. Normally the
|
|
full scale is -1.0 to +1.0. This is an unscaled FFT and for a sine
|
|
wave centered in frequency on a bin and of FS amplitude, the power
|
|
at that center bin will grow by 1024^2/4 = 262144 without windowing.
|
|
Windowing loss cuts this down. The RMS level can grow to sqrt(262144)
|
|
or 512. The dBFS has been scaled to make this max value 0 dBFS by
|
|
removing 54.2 dB. With floating point, the dynamic range is maintained
|
|
no matter how it is scaled, but scaling needs to be considered
|
|
when building the INO.</p>
|
|
|
|
<p>For a 44.1 kHz sample rate, it takes 2903 microseconds (uSec) to
|
|
collect 128 data points. The sum total of all Audio processing
|
|
must be less than this for every update cycle, or overrun will occur
|
|
with severe consequences. For the FFT, the processing time used per
|
|
udate varies cyclicly. The following times are for the most time
|
|
consuming of the updates, i.e., the "max" value.</p>
|
|
<pre class="desc">
|
|
T3.6 Windowed, Power Out, 682 uSec
|
|
T3.6 Windowed, dBFS out, 834 uSec
|
|
T4.0 Windowed, Power Out, 54 uSec
|
|
T4.0 Windowed, dBFS Out, 203 uSec
|
|
</pre>
|
|
<p>This class was improved in March 2021 by using a single 512-point
|
|
FFT to process the 1024 point "real" input to the FFT. This speeds the
|
|
process and also reduces the memory requirements for the FFT. Input and output
|
|
formats and functions remain unchanged.</p>
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioAnalyzeFFT1024_F32">>
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
|
|
<div>
|
|
<script type="text/x-red" data-help-name="AudioAnalyzeFFT256_IQ_F32">
|
|
<!-- ============ AudioAnalyzeFFT256_IQ_F32 ========= -->
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>Does complex (I-Q) input FFT of 256 points. Output is magnitude
|
|
only in formats of RMS (same as I16 version,
|
|
power or dBFS (full scale). Output can be bin by bin or by a pointer to
|
|
the output array. Multiple windowing options are available.</p>
|
|
</div>
|
|
<h3>Boards Supported</h3>
|
|
<ul>
|
|
<li>Teensy 3.2</li>
|
|
<li>Teensy 3.5</li>
|
|
<li>Teensy 3.6</li>
|
|
<li>Teensy 4.0</li>
|
|
<li>Teensy 4.1</li>
|
|
</ul>
|
|
|
|
<h3>Audio Connections</h3>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=top><th>Port</th><th>Purpose</th></tr>
|
|
<tr class=odd><td align=center>In 0</td><td>I Input Signal</td></tr>
|
|
<tr class=odd><td align=center>In 1</td><td>Q Input Signal</td></tr>
|
|
<tr class=odd><td align=center>Out 0</td><td>Overlap 0, I Signal</td></tr>
|
|
<tr class=odd><td align=center>Out 1</td><td>Overlap 0, Q Signal</td></tr>
|
|
<tr class=odd><td align=center>Out 2</td><td>Overlap 1, I Signal</td></tr>
|
|
<tr class=odd><td align=center>Out 3</td><td>Overlap 1, Q Signal</td></tr>
|
|
</table>
|
|
<p>Note: Audio outputs are not yet implemented. RMS, Power and dBFS outputs via
|
|
functions, below, are fully implemented.</p>
|
|
|
|
<h3>Functions</h3>
|
|
<p class=func><span class=keyword>available</span>();</p>
|
|
<p class=desc>returns <strong>bool</strong> true if the FFT is complete,
|
|
otherwise returns false.</p>
|
|
|
|
<p class=func><span class=keyword>read</span>(<strong>int</strong> nBin);</p>
|
|
<p class=desc>Returns the output level for the specified nBin (0, 255).
|
|
Bin 0 is DC and the bins are spaced at the sampling frequency divided
|
|
by 256 Hz.</p>
|
|
|
|
<p class=func><span class=keyword>read</span>(<strong>int</strong> nBinFirst, <strong>int</strong> nBinLast);</p>
|
|
<p class=desc>Returns the power sum for the specified range of bins
|
|
numbered (0, 255). This has the effect of creating a new bin with greater
|
|
width.</p>
|
|
|
|
<p class=func><span class=keyword>windowFunction</span>(<strong>int</strong> win);</p>
|
|
<p class=desc>Sets the windowing function. Instead of calling by number,
|
|
these can be called by the following defined names: </p>
|
|
<pre class="desc">
|
|
AudioWindowNone
|
|
AudioWindowHanning256
|
|
AudioWindowBlackmanHarris256
|
|
</pre>
|
|
|
|
<p class=func><span class=keyword>windowFunction</span>(AudioWindowKaiser256, <strong>float</strong> kdb);</p>
|
|
<p class=desc>Sets the Kaiser window with the first sidelobe kdb
|
|
below the peak. The sidelobes continue to drop going away from a sine-wave
|
|
carrier. This is a very flexible and useful windowing function. </p>
|
|
|
|
<p class=func><span class=keyword>putWindow</span>(<strong>float</strong> *pWin);</p>
|
|
<p class=desc>Activates an INO provided window of 256 float numbers. This replaces
|
|
any window from WindowFunction(). pWin is a pointer to the INO provided window array.
|
|
|
|
<p class=func><span class=keyword>getData</span>();</p>
|
|
<p class=desc>Returns a pointer to an array of 256 float outputs. This
|
|
can save 255 calls to read(). The data remains for about 10 milliseconds.</p>
|
|
|
|
<p class=func><span class=keyword>getWindow</span>();</p>
|
|
<p class=desc>Returns a pointer to an array of 256 floating point numbers
|
|
that is the windowing function in use. This is mostly for diagnostics,
|
|
and not normally needed.</p>
|
|
|
|
<p class=func><span class=keyword>setOutputType</span>(<strong>int</strong> nType);</p>
|
|
<p class=desc>Selects the output form, for example, myFFT.setOutputType(FFT_DBFS); The
|
|
options are</p>
|
|
<pre class="desc">
|
|
FFT_RMS
|
|
FFT_POWER
|
|
FFT_DBFS
|
|
</pre>
|
|
|
|
<p class=func><span class=keyword>setNAverage</span>(<strong>int</strong> nAverage);</p>
|
|
<p class=desc>Selects number of FFT outputs that are power averaged before
|
|
becoming available. This "non-coherent integration."</p>
|
|
|
|
<p class=func><span class=keyword>setXAxis</span>(<strong>uint8_t</strong> xAxis);</p>
|
|
<p class=desc>Re arranges the output order of the frequencies corresponding
|
|
to the various bins. The least significant 2 bit are used. For sin input to
|
|
I and cosine input to Q, the following apply:
|
|
<pre class="desc">
|
|
If xAxis=0 f=fs/2 in middle, f=0 on right edge
|
|
If xAxis=1 f=fs/2 in middle, f=0 on left edge
|
|
If xAxis=2 f=fs/2 on left edge, f=0 in middle
|
|
If xAxis=3 f=fs/2 on right edgr, f=0 in middle
|
|
</pre>
|
|
</p>
|
|
|
|
<h3>Examples</h3>
|
|
<p class=exam>File > Examples > OpenAudio_ArduinoLibrary > TestFFT256iq
|
|
</p>
|
|
|
|
<h3>Notes</h3>
|
|
<p><strong>Scaling - </strong>
|
|
Full scale for floating point DSP is a nebulous concept. Normally the
|
|
full scale is -1.0 to +1.0. This is an unscaled FFT and for a sine
|
|
wave centered in frequency on a bin and of FS amplitude, the power
|
|
at that center bin will grow by 256^2/4 = 65536 without windowing.
|
|
Windowing loss cuts this down. The RMS level can grow to sqrt(65536)
|
|
or 256. The dBFS has been scaled to make this max value 0 dBFS by
|
|
removing 42.1 dB. With floating point, the dynamic range is maintained
|
|
no matter how it is scaled, but scaling needs to be considered
|
|
when building the INO.</p>
|
|
|
|
<p>The only block size supported by this FFT is 128 which is the normal
|
|
default block size.</p>
|
|
|
|
<!--
|
|
<p>For a 44.1 kHz sample rate, it takes 2903 microseconds (uSec) to
|
|
collect 128 data points. The sum total of all Audio processing
|
|
must be less than this for every update cycle, or overrun will occur
|
|
with severe consequences. For the FFT, the processing time used per
|
|
udate varies cyclicly. The following times are for the most time
|
|
consuming of the updates, i.e., the "max" value.</p>
|
|
<pre class="desc">
|
|
T3.6 Windowed, RMS out, - uSec max
|
|
T3.6 Windowed, Power Out, - uSec max
|
|
T3.6 Windowed, dBFS out, - uSec max
|
|
No Window saves 60 uSec on T3.6 for any output.
|
|
T4.0 Windowed, RMS Out, - uSec
|
|
</pre>
|
|
-->
|
|
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioAnalyzeFFT256_IQ_F32">>
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
<div>
|
|
<script type="text/x-red" data-help-name="AudioAnalyzeFFT1024_IQ_F32">
|
|
<!-- ============ AudioAnalyzeFFT1024_IQ_F32 ========= -->
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>Does complex (I-Q) input FFT of 1024 points. Output is magnitude
|
|
only in formats of RMS (same as I16 version,
|
|
power or dBFS (full scale). Output can be bin by bin or by a pointer to
|
|
the output array. Multiple windowing options are available.</p>
|
|
</div>
|
|
<h3>Boards Supported</h3>
|
|
<ul>
|
|
<li>Teensy 3.5</li>
|
|
<li>Teensy 3.6</li>
|
|
<li>Teensy 4.0</li>
|
|
<li>Teensy 4.1</li>
|
|
</ul>
|
|
|
|
<h3>Audio Connections</h3>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=top><th>Port</th><th>Purpose</th></tr>
|
|
<tr class=odd><td align=center>In 0</td><td>I Input Signal</td></tr>
|
|
<tr class=odd><td align=center>In 1</td><td>Q Input Signal</td></tr>
|
|
<tr class=odd><td align=center>Out 0</td><td>Overlap 0, I Signal</td></tr>
|
|
<tr class=odd><td align=center>Out 1</td><td>Overlap 0, Q Signal</td></tr>
|
|
<tr class=odd><td align=center>Out 2</td><td>Overlap 1, I Signal</td></tr>
|
|
<tr class=odd><td align=center>Out 3</td><td>Overlap 1, Q Signal</td></tr>
|
|
</table>
|
|
<p>Note: Audio outputs are not yet implemented. RMS, Power and dBFS outputs via
|
|
functions, below, are fully implemented.</p>
|
|
|
|
<h3>Functions</h3>
|
|
<p class=func><span class=keyword>available</span>();</p>
|
|
<p class=desc>returns <strong>bool</strong> true if the FFT is complete,
|
|
otherwise returns false.</p>
|
|
|
|
<p class=func><span class=keyword>read</span>(<strong>int</strong> nBin);</p>
|
|
<p class=desc>Returns the output level for the specified nBin (0, 1023).
|
|
Bin 0 is DC and the bins are spaced at the sampling frequency divided
|
|
by 1024 Hz.</p>
|
|
|
|
<p class=func><span class=keyword>read</span>(<strong>int</strong> nBinFirst, <strong>int</strong> nBinLast);</p>
|
|
<p class=desc>Returns the power sum for the specified range of bins
|
|
numbered (0, 1023). This has the effect of creating a new bin with greater
|
|
width.</p>
|
|
|
|
<p class=func><span class=keyword>windowFunction</span>(<strong>int</strong> win);</p>
|
|
<p class=desc>Sets the windowing function. Instead of calling by number,
|
|
these can be called by the following defined names: </p>
|
|
<pre class="desc">
|
|
AudioWindowNone
|
|
AudioWindowHanning1024
|
|
AudioWindowBlackmanHarris1024
|
|
</pre>
|
|
|
|
<p class=func><span class=keyword>windowFunction</span>(AudioWindowKaiser1024, <strong>float</strong> kdb);</p>
|
|
<p class=desc>Sets the Kaiser window with the first sidelobe kdb
|
|
below the peak. The sidelobes continue to drop going away from a sine-wave
|
|
carrier. This is a very flexible and useful windowing function. </p>
|
|
|
|
<p class=func><span class=keyword>putWindow</span>(<strong>float</strong> *pWin);</p>
|
|
<p class=desc>Activates an INO provided window of 1024 float numbers. This replaces
|
|
any window from WindowFunction(). pWin is a pointer to the INO provided window array.
|
|
|
|
<p class=func><span class=keyword>getData</span>();</p>
|
|
<p class=desc>Returns a pointer to an array of 1024 float outputs. This
|
|
can save 255 calls to read(). The data remains for about 10 milliseconds.</p>
|
|
|
|
<p class=func><span class=keyword>getWindow</span>();</p>
|
|
<p class=desc>Returns a pointer to an array of 1024 floating point numbers
|
|
that is the windowing function in use. This is mostly for diagnostics,
|
|
and not normally needed.</p>
|
|
|
|
<p class=func><span class=keyword>setOutputType</span>(<strong>int</strong> nType);</p>
|
|
<p class=desc>Selects the output form, for example, myFFT.setOutputType(FFT_DBFS); The
|
|
options are:
|
|
<pre class="desc">
|
|
FFT_RMS
|
|
FFT_POWER
|
|
FFT_DBFS
|
|
</pre>
|
|
</p>
|
|
|
|
<p class=func><span class=keyword>setNAverage</span>(<strong>int</strong> nAverage);</p>
|
|
<p class=desc>Selects number of FFT outputs that are power averaged before
|
|
becoming available. This "non-coherent integration."</p>
|
|
|
|
<p class=func><span class=keyword>setXAxis</span>(<strong>uint8_t</strong> xAxis);</p>
|
|
<p class=desc>Re arranges the output order of the frequencies corresponding
|
|
to the various bins. The least significant 2 bit are used. For sin input to
|
|
I and cosine input to Q, the following apply:
|
|
<pre class="desc">
|
|
If xAxis=0 f=fs/2 in middle, f=0 on right edge
|
|
If xAxis=1 f=fs/2 in middle, f=0 on left edge
|
|
If xAxis=2 f=fs/2 on left edge, f=0 in middle
|
|
If xAxis=3 f=fs/2 on right edgr, f=0 in middle
|
|
</pre>
|
|
</p>
|
|
|
|
<h3>Examples</h3>
|
|
<p class=exam>File > Examples > OpenAudio_ArduinoLibrary > TestFFT1024iq
|
|
</p>
|
|
|
|
<h3>Notes</h3>
|
|
<p><strong>Scaling - </strong>
|
|
Full scale for floating point DSP is a nebulous concept. Normally the
|
|
full scale is -1.0 to +1.0. This is an unscaled FFT and for a sine
|
|
wave centered in frequency on a bin and of FS amplitude, the power
|
|
at that center bin will grow by 1024^2/4 = 262144 without windowing.
|
|
Windowing loss cuts this down some. The RMS level can grow to sqrt(262144)
|
|
or 512. The dBFS has been scaled to make this max value 0 dBFS by
|
|
removing 54.2 dB. With floating point, the dynamic range is maintained
|
|
no matter how it is scaled, but scaling needs to be considered
|
|
when building the INO.</p>
|
|
|
|
<p>The only block size supported by this FFT is 128 which is the normal
|
|
default block size.</p>
|
|
|
|
<!--
|
|
<p>For a 44.1 kHz sample rate, it takes 2903 microseconds (uSec) to
|
|
collect 128 data points. The sum total of all Audio processing
|
|
must be less than this for every update cycle, or overrun will occur
|
|
with severe consequences. For the FFT, the processing time used per
|
|
udate varies cyclicly. The following times are for the most time
|
|
consuming of the updates, i.e., the "max" value.</p>
|
|
<pre class="desc">
|
|
T3.6 Windowed, RMS out, - uSec max
|
|
T3.6 Windowed, Power Out, - uSec max
|
|
T3.6 Windowed, dBFS out, - uSec max
|
|
No Window saves 60 uSec on T3.6 for any output.
|
|
T4.0 Windowed, RMS Out, - uSec
|
|
</pre>
|
|
-->
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioAnalyzeFFT1024_IQ_F32">>
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
|
|
<div>
|
|
<script type="text/x-red" data-help-name="AudioAnalyzeFFT2048_IQ_F32">
|
|
<!-- ============ AudioAnalyzeFFT2048_IQ_F32 ========= -->
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>Does complex (I-Q) input FFT of 2048 points. Output is magnitude
|
|
only in formats of RMS (same as I16 version,
|
|
power or dBFS (full scale). Output can be bin by bin or by a pointer to
|
|
the output array. Multiple windowing options are available.</p>
|
|
</div>
|
|
<h3>Boards Supported</h3>
|
|
<p><strong>Note: Teensy 3.x will NOT compile or run this class.</strong></p>
|
|
<ul>
|
|
<li>Teensy 4.0</li>
|
|
<li>Teensy 4.1</li>
|
|
</ul>
|
|
|
|
<h3>Audio Connections</h3>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=top><th>Port</th><th>Purpose</th></tr>
|
|
<tr class=odd><td align=center>In 0</td><td>I Input Signal</td></tr>
|
|
<tr class=odd><td align=center>In 1</td><td>Q Input Signal</td></tr>
|
|
<tr class=odd><td align=center>Out 0</td><td>Overlap 0, I Signal</td></tr>
|
|
<tr class=odd><td align=center>Out 1</td><td>Overlap 0, Q Signal</td></tr>
|
|
<tr class=odd><td align=center>Out 2</td><td>Overlap 1, I Signal</td></tr>
|
|
<tr class=odd><td align=center>Out 3</td><td>Overlap 1, Q Signal</td></tr>
|
|
</table>
|
|
<p>Note: Audio outputs are not yet implemented. RMS, Power and dBFS outputs via
|
|
functions, below, are fully implemented.</p>
|
|
|
|
<h3>Functions</h3>
|
|
<p class=func><span class=keyword>available</span>();</p>
|
|
<p class=desc>returns <strong>bool</strong> true if the FFT is complete,
|
|
otherwise returns false.</p>
|
|
|
|
<p class=func><span class=keyword>read</span>(<strong>int</strong> nBin);</p>
|
|
<p class=desc>Returns the output level for the specified nBin (0, 2047).
|
|
Bin 0 is DC and the bins are spaced at the sampling frequency divided
|
|
by 2048 Hz.</p>
|
|
|
|
<p class=func><span class=keyword>read</span>(<strong>int</strong> nBinFirst, <strong>int</strong> nBinLast);</p>
|
|
<p class=desc>Returns the power sum for the specified range of bins
|
|
numbered (0, 2047). This has the effect of creating a new bin with greater
|
|
width.</p>
|
|
|
|
<p class=func><span class=keyword>windowFunction</span>(<strong>int</strong> win);</p>
|
|
<p class=desc>Sets the windowing function. Instead of calling by number,
|
|
these can be called by the following defined names: </p>
|
|
<pre class="desc">
|
|
AudioWindowNone
|
|
AudioWindowHanning2048
|
|
AudioWindowBlackmanHarris2048
|
|
</pre>
|
|
|
|
<p class=func><span class=keyword>windowFunction</span>(AudioWindowKaiser2048, <strong>float</strong> kdb);</p>
|
|
<p class=desc>Sets the Kaiser window with the first sidelobe kdb
|
|
below the peak. The sidelobes continue to drop going away from a sine-wave
|
|
carrier. This is a very flexible and useful windowing function. </p>
|
|
|
|
<p class=func><span class=keyword>putWindow</span>(<strong>float</strong> *pWin);</p>
|
|
<p class=desc>Activates an INO provided window of 2048 float numbers. This replaces
|
|
any window from WindowFunction(). pWin is a pointer to the INO provided window array.
|
|
|
|
<p class=func><span class=keyword>getData</span>();</p>
|
|
<p class=desc>Returns a pointer to an array of 2048 float outputs. This
|
|
can save 255 calls to read(). The data remains for about 10 milliseconds.</p>
|
|
|
|
<p class=func><span class=keyword>getWindow</span>();</p>
|
|
<p class=desc>Returns a pointer to an array of 2048 floating point numbers
|
|
that is the windowing function in use. This is mostly for diagnostics,
|
|
and not normally needed.</p>
|
|
|
|
<p class=func><span class=keyword>setOutputType</span>(<strong>int</strong> nType);</p>
|
|
<p class=desc>Selects the output form, for example, myFFT.setOutputType(FFT_DBFS); The
|
|
options are:
|
|
<pre class="desc">
|
|
FFT_RMS
|
|
FFT_POWER
|
|
FFT_DBFS
|
|
</pre>
|
|
</p>
|
|
|
|
<p class=func><span class=keyword>setNAverage</span>(<strong>int</strong> nAverage);</p>
|
|
<p class=desc>Selects number of FFT outputs that are power averaged before
|
|
becoming available. This "non-coherent integration."</p>
|
|
|
|
<p class=func><span class=keyword>setXAxis</span>(<strong>uint8_t</strong> xAxis);</p>
|
|
<p class=desc>Re arranges the output order of the frequencies corresponding
|
|
to the various bins. The least significant 2 bit are used. For sin input to
|
|
I and cosine input to Q, the following apply:
|
|
<pre class="desc">
|
|
If xAxis=0 f=fs/2 in middle, f=0 on right edge
|
|
If xAxis=1 f=fs/2 in middle, f=0 on left edge
|
|
If xAxis=2 f=fs/2 on left edge, f=0 in middle
|
|
If xAxis=3 f=fs/2 on right edgr, f=0 in middle
|
|
</pre>
|
|
</p>
|
|
|
|
<h3>Examples</h3>
|
|
<p class=exam>File > Examples > OpenAudio_ArduinoLibrary > TestFFT2048iq
|
|
</p>
|
|
|
|
<h3>Notes</h3>
|
|
<p><strong>Scaling - </strong>
|
|
Full scale for floating point DSP is a nebulous concept. Normally the
|
|
full scale is -1.0 to +1.0. This is an unscaled FFT and for a sine
|
|
wave centered in frequency on a bin and of FS amplitude, the power
|
|
at that center bin will grow by about a million.
|
|
The dBFS has been scaled to make this max value 0 dBFS by
|
|
removing 60.2 dB. With floating point, the dynamic range is maintained
|
|
no matter how it is scaled, but scaling needs to be considered
|
|
when building the INO.</p>
|
|
|
|
<p>The only block size supported by this FFT is 128 which is the normal
|
|
default block size. Any sampling rate can be supported,
|
|
within maximum available time constraints.</p>
|
|
|
|
<p>For a 44.1 kHz sample rate, it takes 2903 microseconds (uSec) to
|
|
collect 128 data points. The sum total of all Audio processing
|
|
must be less than this for every update cycle, or overrun will occur
|
|
with severe consequences. For the FFT, the processing time used per
|
|
udate varies cyclicly. The Teensy 4.x was measured at 987 microseconds
|
|
per update with windowing and dBFS output. This can probably run with
|
|
96 kHz sample rates, but not 192 kHz.</p>
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioAnalyzeFFT2048_IQ_F32">>
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
|
|
<div>
|
|
<script type="text/x-red" data-help-name="AudioAnalyzeFFT4096_IQ_F32">
|
|
<!-- ============ AudioAnalyzeFFT4096_IQ_F32 ========= -->
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>Does complex (I-Q) input FFT of 4096 points. Output is magnitude
|
|
only in formats of RMS, power or dBFS (full scale).
|
|
Output can be bin by bin or by a pointer to
|
|
the output array. Multiple windowing options are available.</p>
|
|
</div>
|
|
<h3>Boards Supported</h3>
|
|
<p><strong>Note: Teensy 3.x will NOT compile or run this class.</strong></p>
|
|
<ul>
|
|
<li>Teensy 4.0</li>
|
|
<li>Teensy 4.1</li>
|
|
</ul>
|
|
|
|
<h3>Audio Connections</h3>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=top><th>Port</th><th>Purpose</th></tr>
|
|
<tr class=odd><td align=center>In 0</td><td>I Input Signal</td></tr>
|
|
<tr class=odd><td align=center>In 1</td><td>Q Input Signal</td></tr>
|
|
</table>
|
|
|
|
<h3>Functions</h3>
|
|
<p class=func><span class=keyword>available</span>();</p>
|
|
<p class=desc>returns <strong>bool</strong> true if the FFT is complete,
|
|
otherwise returns false.</p>
|
|
|
|
<p class=func><span class=keyword>read</span>(<strong>int</strong> nBin);</p>
|
|
<p class=desc>Returns the output level for the specified nBin (0, 4095).
|
|
Bin 0 is DC and the bins are spaced at the sampling frequency divided
|
|
by 4096 Hz.</p>
|
|
|
|
<p class=func><span class=keyword>read</span>(<strong>int</strong> nBinFirst, <strong>int</strong> nBinLast);</p>
|
|
<p class=desc>Returns the power sum for the specified range of bins
|
|
numbered (0, 4095). This has the effect of creating a new bin with greater
|
|
width.</p>
|
|
|
|
<p class=func><span class=keyword>windowFunction</span>(<strong>int</strong> win);</p>
|
|
<p class=desc>Sets the windowing function. Instead of calling by number,
|
|
these can be called by the following defined names: </p>
|
|
<pre class="desc">
|
|
AudioWindowNone
|
|
AudioWindowHanning4096
|
|
AudioWindowBlackmanHarris4096
|
|
</pre>
|
|
|
|
<p class=func><span class=keyword>windowFunction</span>(AudioWindowKaiser4096, <strong>float</strong> kdb);</p>
|
|
<p class=desc>Sets the Kaiser window with the first sidelobe kdb
|
|
below the peak. The sidelobes continue to drop going away from a sine-wave
|
|
carrier. This is a very flexible and useful windowing function. </p>
|
|
|
|
<p class=func><span class=keyword>putWindow</span>(<strong>float</strong> *pWin);</p>
|
|
<p class=desc>Activates an INO provided window of 4096 float numbers. This replaces
|
|
any window from WindowFunction(). pWin is a pointer to the INO provided window array.
|
|
|
|
<p class=func><span class=keyword>getData</span>();</p>
|
|
<p class=desc>Returns a pointer to an array of 4096 float outputs. This
|
|
can save 255 calls to read(). The data remains for about 10 milliseconds.</p>
|
|
|
|
<p class=func><span class=keyword>getWindow</span>();</p>
|
|
<p class=desc>Returns a pointer to an array of 4096 floating point numbers
|
|
that is the windowing function in use. This is mostly for diagnostics,
|
|
and not normally needed.</p>
|
|
|
|
<p class=func><span class=keyword>setOutputType</span>(<strong>int</strong> nType);</p>
|
|
<p class=desc>Selects the output form, for example, myFFT.setOutputType(FFT_DBFS); The
|
|
options are:
|
|
<pre class="desc">
|
|
FFT_RMS
|
|
FFT_POWER
|
|
FFT_DBFS
|
|
</pre>
|
|
</p>
|
|
|
|
<p class=func><span class=keyword>setNAverage</span>(<strong>int</strong> nAverage);</p>
|
|
<p class=desc>Selects number of FFT outputs that are power averaged before
|
|
becoming available. This "non-coherent integration."</p>
|
|
|
|
<p class=func><span class=keyword>setXAxis</span>(<strong>uint8_t</strong> xAxis);</p>
|
|
<p class=desc>Re arranges the output order of the frequencies corresponding
|
|
to the various bins. The least significant 2 bit are used. For sin input to
|
|
I and cosine input to Q, the following apply:
|
|
<pre class="desc">
|
|
If xAxis=0 f=fs/2 in middle, f=0 on right edge
|
|
If xAxis=1 f=fs/2 in middle, f=0 on left edge
|
|
If xAxis=2 f=fs/2 on left edge, f=0 in middle
|
|
If xAxis=3 f=fs/2 on right edgr, f=0 in middle
|
|
</pre>
|
|
</p>
|
|
|
|
<h3>Examples</h3>
|
|
<p class=exam>File > Examples > OpenAudio_ArduinoLibrary > TestFFT4096iq
|
|
</p>
|
|
|
|
<h3>Notes</h3>
|
|
<p><strong>Memory needs - </strong>As FFTs get bigger they need more memory. This
|
|
4096 point FFT needs 64 F32 memories. Be sure to allocate more than this
|
|
in your INO sketch. In addition, at linking time, 98 kByte of RAM is added.
|
|
The T4.x can support this, but it is a major memory user. </p>
|
|
|
|
<p><strong>Scaling - </strong>
|
|
Full scale for floating point DSP is a nebulous concept. Normally the
|
|
full scale is -1.0 to +1.0. This is an unscaled FFT and for a sine
|
|
wave centered in frequency on a bin and of FS amplitude, the power
|
|
at that center bin will grow by about a million.
|
|
The dBFS has been scaled to make this max value 0 dBFS by
|
|
removing 66.2 dB. With floating point, the dynamic range is maintained
|
|
no matter how it is scaled, but scaling needs to be considered
|
|
when building the INO.</p>
|
|
|
|
<p>The only block size supported by this FFT is 128 which is the normal
|
|
default block size. Any sampling rate can be supported,
|
|
within maximum available time constraints.</p>
|
|
|
|
<p>For a 44.1 kHz sample rate, it takes 2903 microseconds (uSec) to
|
|
collect 128 data points. The sum total of all Audio processing
|
|
must be less than this for every update cycle, or overrun will occur
|
|
with severe consequences. For the FFT, the processing time used per
|
|
udate varies cyclicly. The Teensy 4.x was measured at 710 microseconds
|
|
per update with windowing and dBFS output. This can run with
|
|
96 kHz sample rates, but not 192 kHz. By using FFT_POWER output the maximum
|
|
processor time per update() is only 510 microseconds and, depending on other
|
|
processing, 192 kHz sample rate could be possible.</p>
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioAnalyzeFFT4096_IQ_F32">>
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
<div>
|
|
<script type="text/x-red" data-help-name="AudioAnalyzeFFT4096_IQem_F32">
|
|
<!-- ============ AudioAnalyzeFFT4096_IQem_F32 ========= -->
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>Does complex (I-Q) input FFT of 4096 points. Output is magnitude
|
|
only in formats of RMS, power or dBFS (full scale).
|
|
Output can be bin by bin. Multiple windowing options are available.
|
|
This EM version obtains all memory arrays from the INO.</p>
|
|
</div>
|
|
<p><strong>As of 20 Feb 2022 this is Beta Test and changes may occur in the structure of the calls.</strong></p>
|
|
<h3>Boards Supported</h3>
|
|
<p><strong>Note: Teensy 3.x will NOT compile or run this class.</strong></p>
|
|
<ul>
|
|
<li>Teensy 4.0</li>
|
|
<li>Teensy 4.1</li>
|
|
</ul>
|
|
|
|
<h3>Audio Connections</h3>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=top><th>Port</th><th>Purpose</th></tr>
|
|
<tr class=odd><td align=center>In 0</td><td>I Input Signal</td></tr>
|
|
<tr class=odd><td align=center>In 1</td><td>Q Input Signal</td></tr>
|
|
</table>
|
|
|
|
<h3>Functions</h3>
|
|
<p class=func><span class=keyword>available</span>();</p>
|
|
<p class=desc>returns <strong>bool</strong> true if the FFT is complete,
|
|
otherwise returns false.</p>
|
|
|
|
<p class=func><span class=keyword>read</span>(<strong>int</strong> nBin);</p>
|
|
<p class=desc>Returns the output level for the specified nBin (0, 4095).
|
|
Bin 0 is DC and the bins are spaced at the sampling frequency divided
|
|
by 4096 in Hz.</p>
|
|
|
|
<p class=func><span class=keyword>read</span>(<strong>int</strong> nBinFirst, <strong>int</strong> nBinLast);</p>
|
|
<p class=desc>Returns the power sum for the specified range of bins
|
|
numbered (0, 4095). This has the effect of creating a new bin with greater
|
|
width.</p>
|
|
|
|
<p class=func><span class=keyword>windowFunction</span>(<strong>int</strong> win);</p>
|
|
<p class=desc>Sets the windowing function. Instead of calling by number,
|
|
these can be called by the following defined names: </p>
|
|
<pre class="desc">
|
|
AudioWindowNone
|
|
AudioWindowHanning4096
|
|
AudioWindowBlackmanHarris4096
|
|
</pre>
|
|
|
|
<p class=func><span class=keyword>windowFunction</span>(AudioWindowKaiser4096, <strong>float</strong> kdb);</p>
|
|
<p class=desc>Sets the Kaiser window with the first sidelobe kdb
|
|
below the peak. The sidelobes continue to drop going away from a sine-wave
|
|
carrier. This is a very flexible and useful windowing function. </p>
|
|
|
|
<p class=func><span class=keyword>setOutputType</span>(<strong>int</strong> nType);</p>
|
|
<p class=desc>Selects the output form, for example, myFFT.setOutputType(FFT_DBFS); The
|
|
options are:
|
|
<pre class="desc">
|
|
FFT_RMS
|
|
FFT_POWER
|
|
FFT_DBFS
|
|
</pre>
|
|
</p>
|
|
|
|
<p class=func><span class=keyword>setNAverage</span>(<strong>int</strong> nAverage);</p>
|
|
<p class=desc>Selects number of FFT outputs that are power averaged before
|
|
becoming available. This "non-coherent integration."</p>
|
|
|
|
<p class=func><span class=keyword>setXAxis</span>(<strong>uint8_t</strong> xAxis);</p>
|
|
<p class=desc>Re arranges the output order of the frequencies corresponding
|
|
to the various bins. The least significant 2 bit are used. For sin input to
|
|
I and cosine input to Q, the following apply:
|
|
<pre class="desc">
|
|
If xAxis=0 f=0 in middle, f=fs/2 on left edge
|
|
If xAxis=1 f=0 in middle, f=fs/2 on right edge
|
|
If xAxis=2 f=0 on right edge, f=fs/2 in middle
|
|
If xAxis=3 f=0 on left edge, f=fs/2 in middle
|
|
</pre>
|
|
</p>
|
|
|
|
<h3>Examples</h3>
|
|
<p class=exam>File > Examples > OpenAudio_ArduinoLibrary > TestFFT4096iqEM
|
|
</p>
|
|
|
|
<h3>Notes</h3>
|
|
<p><strong>Memory needs - </strong>As FFTs get bigger they need more memory. This
|
|
4096 point FFT needs 64 F32 memories. Be sure to allocate more than this
|
|
in your INO sketch. In addition, at linking time, 58 kByte of .INO supplied
|
|
RAM is added without use of power averaging and 74 kBytes if it is used.
|
|
The T4.x can support this, but it is a major memory user. </p>
|
|
|
|
<p><strong>Scaling - </strong>
|
|
Full scale for floating point DSP is a nebulous concept. Normally the
|
|
full scale is -1.0 to +1.0. This is an unscaled FFT and for a sine
|
|
wave centered in frequency on a bin and of FS amplitude, the power
|
|
at that center bin will grow by about a million.
|
|
The dBFS has been scaled to make this max value 0 dBFS by
|
|
removing 66.2 dB. With floating point, the dynamic range is maintained
|
|
no matter how it is scaled, but scaling needs to be considered
|
|
when building the INO.</p>
|
|
|
|
<p>The only block size supported by this FFT is 128 which is the normal
|
|
default block size. Any sampling rate can be supported,
|
|
within maximum available time constraints.</p>
|
|
|
|
<p>For a 44.1 kHz sample rate, it takes 2903 microseconds (uSec) to
|
|
collect 128 data points. The sum total of all Audio processing
|
|
must be less than this for every update cycle, or overrun will occur
|
|
with severe consequences. For the FFT, the processing time used per
|
|
udate varies cyclicly. The Teensy 4.x was measured at 710 microseconds
|
|
per update with windowing and dBFS output. This can run with
|
|
96 kHz sample rates, but not 192 kHz. By using FFT_POWER output the maximum
|
|
processor time per update() is only 510 microseconds and, depending on other
|
|
processing, 192 kHz sample rate could be possible.</p>
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioAnalyzeFFT4096em_IQ_F32">>
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
|
|
|
|
<div>
|
|
<script type="text/x-red" data-help-name="AudioAnalyzeToneDetect_F32">
|
|
<!-- ============ AudioAnalyzeToneDetect_F32 ========= -->
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>Parallels the Teensy Audio class that detects sine wave tones. Uses the Goertzel
|
|
algorithm and has programmable tone detection time.</p>
|
|
</div>
|
|
<h3>Boards Supported</h3>
|
|
<ul>
|
|
<li>Teensy 3.2</li>
|
|
<li>Teensy 3.5</li>
|
|
<li>Teensy 3.6</li>
|
|
<li>Teensy 4.0</li>
|
|
<li>Teensy 4.1</li>
|
|
</ul>
|
|
<h3>Audio Connections</h3>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=top><th>Port</th><th>Purpose</th></tr>
|
|
<tr class=odd><td align=center>In 0</td><td>I Input Signal</td></tr>
|
|
</table>
|
|
|
|
<h3>Functions</h3>
|
|
<p class=func><span class=keyword>frequency</span>(<strong>float</strong> freq, <strong>int</strong> cycles);</p>
|
|
<p class=desc>Sets the frequency fr for tone detection, in Hz, and the number of cycles to be analyzed.</p>
|
|
|
|
<p class=func><span class=keyword>available</span>();</p>
|
|
<p class=desc>returns <strong>bool</strong> true if the tone detection is complete,
|
|
otherwise returns false.</p>
|
|
|
|
<p class=func><span class=keyword>read</span>();</p>
|
|
<p class=desc>Returns the output rms level.</p>
|
|
|
|
<p class=func><span class=keyword>threshold</span>(<strong>float</strong> thresh);</p>
|
|
<p class=desc>Sets the threshold for the bool() function. Range of 0.0 to 1.0.</p>
|
|
|
|
<p class=func><span class=keyword>bool</span>();</p>
|
|
<p class=desc>Returns true if rms level is above threshold..</p>
|
|
|
|
<h3>Examples</h3>
|
|
<p class=exam>File > Examples > OpenAudio_ArduinoLibrary > ToneDetect1
|
|
</p>
|
|
|
|
<h3>Notes</h3>
|
|
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioAnalyzeToneDetect_F32">>
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
<div>
|
|
<script type="text/x-red" data-help-name="AudioAnalyzeCTCSS_F32">
|
|
<!-- ============ AudioAnalyzeCTCSS_F32 ========= -->
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>Specific to the CTCSS sub-audible tone detection. Uses the Goertzel
|
|
algorithm, can be programmed for any tone in the 67 to 254 Hz range.
|
|
See analyze_CTCSS_F32.h for much information.
|
|
</p>
|
|
</div>
|
|
<h3>Boards Supported</h3>
|
|
<ul>
|
|
<li>Teensy 3.2</li>
|
|
<li>Teensy 3.5</li>
|
|
<li>Teensy 3.6</li>
|
|
<li>Teensy 4.0</li>
|
|
<li>Teensy 4.1</li>
|
|
</ul>
|
|
<h3>Audio Connections</h3>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=top><th>Port</th><th>Purpose</th></tr>
|
|
<tr class=odd><td align=center>In 0</td><td>I Input Signal</td></tr>
|
|
<tr class=odd><td align=center>Out 0</td><td>O Output Signal</td></tr>
|
|
</table>
|
|
|
|
<h3>Functions</h3>
|
|
<p class=func><span class=keyword>initCTCSS</span>(<strong>void</strong>);</p>
|
|
<p class=desc>Run this to initialize the CTCSS tone detector This function is <strong>required.</strong>
|
|
|
|
<p class=func><span class=keyword>frequency</span>(<strong>float</strong> freq, <strong>int</strong> tMeas);</p>
|
|
<p class=desc>Sets the frequency freq for tone detection, in Hz between
|
|
67.0 and 254.1 Hz, and the number of milliseconds to be analyzed. The defaults are
|
|
103.5 Hz and 300 milliseconds. The parameter tMeas is optional.</p>
|
|
|
|
<p class=func><span class=keyword>readTonePower</span>(<strong>void</strong>);</p>
|
|
<p class=desc>Returns the power measured for the narrow-band Goertzel tone filter as a <strong>float</strong>.</p>
|
|
|
|
<p class=func><span class=keyword>readRefPower</span>(<strong>void</strong>);</p>
|
|
<p class=desc>Returns the power measured for the 67 to 254 Hz filter as a <strong>float</strong>. The
|
|
tone frequency is notched out for this measurement.</p>
|
|
|
|
|
|
<p class=func><span class=keyword>readTonePresent</span>(<strong>uint16_t</strong> what);</p>
|
|
<p class=desc>Returns a <strong>bool</strong> true or false to indicate whether thecurrent reading
|
|
is above or below the threshold condition set by "what". Values for <strong>uint16_t</strong>
|
|
parameter "what" are the following pre-defined values:</p>
|
|
<pre class="desc">
|
|
isAbsThreshold
|
|
isRelThreshold
|
|
isBothThreshold
|
|
</pre>
|
|
|
|
<p class=func><span class=keyword>thresholds</span>(<strong>float</strong> levelAbs, <strong>float</strong> levelRel);</p>
|
|
<p class=desc>The parameter levelAbs sets the threshold for comparing pTone with to estimate the presence
|
|
of a CTCSS tone. The parameter levelRel sets a similar threshold on the quotient PowerTone/PowerRef.
|
|
Setting either threshold to 0.0f disables that threshold test.</p>
|
|
|
|
<p class=func><span class=keyword>available</span>(<strong>void</strong>);</p>
|
|
<p class=desc>returns <strong>bool</strong> true if the tone detection is complete,
|
|
otherwise returns false.</p>
|
|
|
|
<p class=func><span class=keyword>setCTCSS_BP</span>(<strong>float*</strong> filterCoeffs);</p>
|
|
<p class=desc>This function sets the IIR coefficients for the 67 to 254 Hz bandpass filter.
|
|
Set filterCoeffs to NULL to use pre-determined coefficients for 44, 48,96 or 100 KHz.
|
|
Alternatively build your own using info in analyze_CTCSS_F32.h. </p>
|
|
|
|
<h3>Examples</h3>
|
|
<p class=exam>File > Examples > OpenAudio_ArduinoLibrary > ToneDetect3
|
|
</p>
|
|
|
|
<h3>Notes</h3>
|
|
<p>Two outputs are available, TonePower and RefPower. TonePower is the output
|
|
of the Goertzel sharply tuned filter and is the conventional CTCSS output
|
|
that is compared to a threshold. RefPower measures the power in the entire
|
|
67 to 254 Hz sub-audible band, except at the tone frequency. </p>
|
|
|
|
<p>Measurements are repeated every tMeas milliseconds, automatically
|
|
and decisions of tone presence are updated.</p>
|
|
|
|
<p>Each update of an 128-input block takes about 42 uSec on a Teensy 3.6.</p>
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioAnalyzeCTCSS_F32">>
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
<!-- ============ AudioAnalyzePeak_F32 ========= -->
|
|
<script type="text/x-red" data-help-name="AudioAnalyzePeak_F32">
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>Track the signal peak amplitude. Very useful for simple
|
|
audio level response projects, and general troubleshooting.
|
|
Almost same class as in Teensy Library but this uses F32 floating point audio input.</p>
|
|
</div>
|
|
<h3>Audio Connections</h3>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=top><th>Port</th><th>Purpose</th></tr>
|
|
<tr class=odd><td align=center>In 0</td><td>Signal to analyze</td></tr>
|
|
</table>
|
|
<h3>Functions</h3>
|
|
<p class=func><span class=keyword>available</span>();</p>
|
|
<p class=desc>Returns true each time new peak data is available.
|
|
</p>
|
|
<p class=func><span class=keyword>read</span>();</p>
|
|
<p class=desc>Read the highest peak amplitude value since the last read.
|
|
</p>
|
|
<p class=func><span class=keyword>readPeakToPeak</span>();</p>
|
|
<p class=desc>Read the highest peak-to-peak amplitude since the last read.
|
|
</p>
|
|
<h3>Examples</h3>
|
|
<p class=exam>File > Examples > OpenAudio_ArduinoLibrary > AudioTestPeakRMS
|
|
</p>
|
|
|
|
<h3>Notes</h3>
|
|
<p>With floating point, F32, audio, there is generally no maximum level that
|
|
represents full scale, as there is with Teensy Audio I16. The
|
|
exception is at the input and output points where there is
|
|
an interface with an integer device. The F32 scaling at these points is set to
|
|
-1.0 and +1.0 corresponding to I16 -32768 and 32767.</p>
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioAnalyzePeak_F32">
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
<!-- ============ AudioAnalyzeRMS_F32 ========= -->
|
|
<script type="text/x-red" data-help-name="AudioAnalyzeRMS_F32">
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>Track the signal RMS amplitude. Useful for
|
|
audio level response projects, and general troubleshooting.
|
|
Almost same class as in Teensy Library but this uses F32 floating point audio input.</p>
|
|
</div>
|
|
<h3>Audio Connections</h3>
|
|
<table class=doc align=center cellpadding=3>
|
|
<tr class=top><th>Port</th><th>Purpose</th></tr>
|
|
<tr class=odd><td align=center>In 0</td><td>Signal to analyze</td></tr>
|
|
</table>
|
|
<h3>Functions</h3>
|
|
<p class=func><span class=keyword>available</span>();</p>
|
|
<p class=desc>Returns true if new RMS data is available.
|
|
</p>
|
|
<p class=func><span class=keyword>read</span>();</p>
|
|
<p class=desc>Read the new RMS value.
|
|
|
|
</p>
|
|
<h3>Examples</h3>
|
|
<p class=exam>File > Examples > OpenAudio_ArduinoLibrary > AudioTestPeakRMS
|
|
</p>
|
|
<h3>Notes</h3>
|
|
<p>With floating point, F32, audio, there is generally no maximum level that
|
|
represents full scale, as there is with Teensy Audio I16. The
|
|
exception is at the input and output points where there is
|
|
an interface with an integer device. The F32 scaling at these points is set to
|
|
-1.0 and +1.0 corresponding to I16 -32768 and 32767.</p>
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioAnalyzeRMS_F32">
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
|
|
<!-- ========== AudioControlSGTL5000 ============= -->
|
|
<script type="text/x-red" data-help-name="AudioControlSGTL5000">
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>Control the SGTL5000 chip on the
|
|
<a href="http://www.pjrc.com/store/teensy3_audio.html" target="_blank">audio shield</a>.
|
|
SGTL5000 is always used in slave mode, where Teensy controls
|
|
all I2S timing.
|
|
</p>
|
|
<p align=center><img src="img/sgtl5000closeup.jpg"></p>
|
|
</div>
|
|
<h3>Audio Connections</h3>
|
|
<p>This object has no audio inputs or outputs. Separate i2s objects
|
|
are used to send and receive audio data. I2S master mode objects
|
|
must be used, because this object configures the SGTL5000 in slave
|
|
mode, where it depends on Teensy to provide all I2S clocks.
|
|
This object controls
|
|
how the SGTL5000 will use those I2S audio streams.</p>
|
|
|
|
<h3>Functions</h3>
|
|
<p>These are the most commonly used SGTL5000 functions.</p>
|
|
<p class=func><span class=keyword>enable</span>();</p>
|
|
<p class=desc>Start the SGTL5000. This function should be called first.
|
|
</p>
|
|
<p class=func><span class=keyword>volume</span>(level);</p>
|
|
<p class=desc>Set the headphone volume level. Range is 0 to 1.0, but
|
|
0.8 corresponds to the maximum undistorted output for a full scale
|
|
signal. Usually 0.5 is a comfortable listening level. The line
|
|
level outputs are <em>not</em> changed by this function.
|
|
</p>
|
|
<p class=func><span class=keyword>inputSelect</span>(input);</p>
|
|
<p class=desc>Select which input to use: AUDIO_INPUT_LINEIN or AUDIO_INPUT_MIC.
|
|
</p>
|
|
<p class=func><span class=keyword>micGain</span>(dB);</p>
|
|
<p class=desc>When using the microphone input, set the amplifier gain.
|
|
The input number is in decibels, from 0 to 63.
|
|
</p>
|
|
|
|
<h3>Signal Levels</h3>
|
|
|
|
<p>The default signal levels should be used for most applications,
|
|
but these functions allow you to customize the analog signals.</p>
|
|
|
|
<p class=func><span class=keyword>muteHeadphone</span>();</p>
|
|
<p class=desc>Silence the headphone output.
|
|
</p>
|
|
<p class=func><span class=keyword>unmuteHeadphone</span>();</p>
|
|
<p class=desc>Turn the headphone output on.
|
|
</p>
|
|
<p class=func><span class=keyword>muteLineout</span>();</p>
|
|
<p class=desc>Silence the line level outputs.
|
|
</p>
|
|
<p class=func><span class=keyword>unmuteLineout</span>();</p>
|
|
<p class=desc>Turn the line level outputs on.
|
|
</p>
|
|
<p class=func><span class=keyword>lineInLevel</span>(both);</p>
|
|
<p class=desc style="padding-bottom:0.2em;">Adjust the sensitivity of the line-level inputs.
|
|
Fifteen settings are possible:
|
|
</p>
|
|
<pre class="desc">
|
|
0: 3.12 Volts p-p
|
|
1: 2.63 Volts p-p
|
|
2: 2.22 Volts p-p
|
|
3: 1.87 Volts p-p
|
|
4: 1.58 Volts p-p
|
|
5: 1.33 Volts p-p (default)
|
|
6: 1.11 Volts p-p
|
|
7: 0.94 Volts p-p
|
|
8: 0.79 Volts p-p
|
|
9: 0.67 Volts p-p
|
|
10: 0.56 Volts p-p
|
|
11: 0.48 Volts p-p
|
|
12: 0.40 Volts p-p
|
|
13: 0.34 Volts p-p
|
|
14: 0.29 Volts p-p
|
|
15: 0.24 Volts p-p
|
|
</pre>
|
|
<p class=func><span class=keyword>lineInLevel</span>(left, right);</p>
|
|
<p class=desc>Adjust the sensitivity of the line-level inputs, with different
|
|
settings for left and right. The same 15 settings are available.
|
|
</p>
|
|
<p class=func><span class=keyword>lineOutLevel</span>(both);</p>
|
|
<p class=desc style="padding-bottom:0.2em;">Adjust the line level output
|
|
voltage range. The following settings are possible:
|
|
</p>
|
|
<pre class="desc">
|
|
13: 3.16 Volts p-p
|
|
14: 2.98 Volts p-p
|
|
15: 2.83 Volts p-p
|
|
16: 2.67 Volts p-p
|
|
17: 2.53 Volts p-p
|
|
18: 2.39 Volts p-p
|
|
19: 2.26 Volts p-p
|
|
20: 2.14 Volts p-p
|
|
21: 2.02 Volts p-p
|
|
22: 1.91 Volts p-p
|
|
23: 1.80 Volts p-p
|
|
24: 1.71 Volts p-p
|
|
25: 1.62 Volts p-p
|
|
26: 1.53 Volts p-p
|
|
27: 1.44 Volts p-p
|
|
28: 1.37 Volts p-p
|
|
29: 1.29 Volts p-p (default)
|
|
30: 1.22 Volts p-p
|
|
31: 1.16 Volts p-p
|
|
</pre>
|
|
<p class=func><span class=keyword>lineOutLevel</span>(left, right);</p>
|
|
<p class=desc>Adjust the line level outout voltage range, with separate
|
|
settings for left and right. The same settings (13 to 31) are available.
|
|
</p>
|
|
|
|
|
|
<h3>Signal Conditioning</h3>
|
|
|
|
<p>Usually these digital signal conditioning features should be left at their
|
|
default settings.
|
|
</p>
|
|
|
|
<p class=func><span class=keyword>adcHighPassFilterFreeze</span>();</p>
|
|
<p class=desc>By default, the analog input (either line-level inputs or mic)
|
|
is high-pass filtered, to remove any DC component. This function
|
|
freezes the filter, so the current DC component is still substracted, but
|
|
the filter stops tracking any DC or low frequency changes.
|
|
</p>
|
|
<p class=func><span class=keyword>adcHighPassFilterDisable</span>();</p>
|
|
<p class=desc>Completely disable the analog input filter. DC and sub-audible
|
|
low frequencies are allowed to enter the digital signal. This
|
|
<a href="http://openaudio.blogspot.com/2017/03/teensy-audio-board-self-noise.html">may
|
|
reduce noise</a> in some cases.
|
|
</p>
|
|
<p class=func><span class=keyword>adcHighPassFilterEnable</span>();</p>
|
|
<p class=desc>Turn the DC-blocking filter back on, if disabled, or
|
|
allows it to resume tracking DC and low frequency changes, if
|
|
previously frozen. This is the default setting.
|
|
</p>
|
|
<p class=func><span class=keyword>dacVolume</span>(both);</p>
|
|
<p class=desc>Normally output volume should be used with volume(), which
|
|
changes the analog gain in the headphone amplifier. This function
|
|
on the other hand controls digital attenuation before conversion to analog, which
|
|
reduces resolution, but allows another fine control of output
|
|
signal level. The ranges is 0 to 1.0, with the default (no digital attenuation)
|
|
at 1.0.
|
|
</p>
|
|
<p class=desc>dacVolume uses zero-crossing detect to avoid clicks, and graceful
|
|
ramping is handled by the chip so that a new volume may be set directly in
|
|
a single call.
|
|
</p>
|
|
<p class=func><span class=keyword>dacVolume</span>(left, right);</p>
|
|
<p class=desc>Adjust the digital output volume separately on left and
|
|
right channels.
|
|
</p>
|
|
<p class=func><span class=keyword>dacVolumeRamp</span>();</p>
|
|
<p class=desc>Enable graceful volume ramping. The dacVolume adjusts gradually using
|
|
an exponential curve. Pops or loud clicks are avoided when making large
|
|
changes in volume level.
|
|
</p>
|
|
<p class=func><span class=keyword>dacVolumeRampLinear</span>();</p>
|
|
<p class=desc>Enable faster volume ramping. A slight click may be heard during a
|
|
large volume change.
|
|
</p>
|
|
<p class=func><span class=keyword>dacVolumeRampDisable</span>();</p>
|
|
<p class=desc>Do not use any gradual ramping. The zero cross feature still helps
|
|
for small changes, but large volume changes may produce a pop or click.
|
|
</p>
|
|
|
|
<h3>Audio Processor</h3>
|
|
|
|
<p>The optional digital audio processor is capable of implementing
|
|
one or more of: automatic volume control, surround sound control,
|
|
bass enhancement, and tonal adjustments (either a
|
|
simple tone control, or a parametric equalizer, or a graphic equalizer),
|
|
in that order.
|
|
</p>
|
|
<p>These signal processing features are implemented in the SGTL5000 chip,
|
|
so they do not consume CPU time on Teensy. However, the order of
|
|
these processes is fixed in the hardware.
|
|
</p>
|
|
<p>It is good practice to mute the outputs before enabling or disabling
|
|
the Audio Processor, to avoid clicks or thumps.
|
|
</p>
|
|
|
|
<p class=func><span class=keyword>audioPreProcessorEnable</span>();</p>
|
|
<p class=desc>Enable the audio processor to pre-process the input
|
|
(from either line-level inputs or microphone) before it's sent
|
|
to Teensy by I2S.
|
|
</p>
|
|
<p class=func><span class=keyword>audioPostProcessorEnable</span>();</p>
|
|
<p class=desc>Enable the audio processor to post-process Teensy's
|
|
I2S output before it's turned into analog signals for the
|
|
headphones and/or line level outputs.
|
|
</p>
|
|
<p class=func><span class=keyword>audioProcessorDisable</span>();</p>
|
|
<p class=desc>Disable the audio processor.
|
|
</p>
|
|
<p class=func><span class=keyword>autoVolumeControl</span>(maxGain, response, hardLimit, threshold, attack, decay);</p>
|
|
<p class=desc>Configures the auto volume control, which is implemented as a compressor/expander
|
|
or hard limiter. <em>maxGain</em> is the maximum gain that can be applied for expanding, and
|
|
can take one of three values: 0 (0dB), 1 (6.0dB) and 2 (12dB). Values greater than 2 are treated
|
|
as 2. <em>response</em> controls the integration time for the compressor and can take
|
|
four values: 0 (0ms), 1 (25ms), 2 (50ms) or 3 (100ms). Larger values average the volume
|
|
over a longer time, allowing short-term peaks through.
|
|
</p>
|
|
<p class=desc>If <em>hardLimit</em> is 0, a 'soft
|
|
knee' compressor is used to progressively compress louder values which are near to or above the
|
|
threashold (the louder they are, the greater the compression). If it is 1, a hard compressor
|
|
is used (all values above the threashold are the same loudness). The <em>threashold</em> is specified
|
|
as a float in the range 0dBFS to -96dBFS, where -18dBFS is a typical value.
|
|
<em>attack</em> is a float controlling the rate of decrease in gain when the signal is over
|
|
threashold, in dB/s. <em>decay</em> controls how fast gain is restored once the level
|
|
drops below threashold, again in dB/s. It is typically set to a longer value than attack.
|
|
</p>
|
|
<p class=func><span class=keyword>autoVolumeEnable</span>();</p>
|
|
<p class=desc>Enables auto volume control, using the previously specified settings.
|
|
</p>
|
|
<p class=func><span class=keyword>autoVolumeDisable</span>();</p>
|
|
<p class=desc>Disables auto volume control.
|
|
</p>
|
|
|
|
<p class=func><span class=keyword>surroundSoundEnable</span>();</p>
|
|
<p class=desc>Enable virtual surround processing, to give a broader and
|
|
deeper stereo image (even with mono input).
|
|
</p>
|
|
<p class=func><span class=keyword>surroundSoundDisable</span>();</p>
|
|
<p class=desc>Disable virtual surround processing. Before disabling, ramp up
|
|
the width to maximum to avoid pops.
|
|
</p>
|
|
<p class=func><span class=keyword>surroundSound</span>(width);</p>
|
|
<p class=desc>Configures virtual surround width from 0 (mono) to 7 (widest).
|
|
</p>
|
|
<p class=func><span class=keyword>surroundSound</span>(width, select);</p>
|
|
<p class=desc>Configures virtual surround width from 0 (mono) to 7 (widest).
|
|
<em>select</em> may be set to 1 (disable), 2 (mono input) or 3 (stereo input).
|
|
</p>
|
|
|
|
<p class=func><span class=keyword>enhanceBassEnable</span>();</p>
|
|
<p class=desc>Enable bass enhancement. A mono, low-pass filtered copy of
|
|
the original stereo signal has bass levels boosted and is then mixed back into
|
|
the stereo signal, which is then optionally high pass filtered (to remove
|
|
inaudible subsonic frequencies).
|
|
</p>
|
|
<p class=func><span class=keyword>enhanceBassDisable</span>();</p>
|
|
<p class=desc>Disable bass enhancement. Before disabling, ramp down the bass
|
|
enhancement level to zero.
|
|
</p>
|
|
<p class=func><span class=keyword>enhanceBass</span>(lr_lev, bass_lev);</p>
|
|
<p class=desc>Configures the bass enhancement by setting the levels of the
|
|
original stereo signal and the bass-enhanced mono level which will be mixed together.
|
|
There is no high-pass filter.
|
|
</p>
|
|
<p class=desc>When changing bass level, call this function repeatedly to ramp up or down the bass in
|
|
steps of 0.5dB, to avoid pops.
|
|
</p>
|
|
<p class=func><span class=keyword>enhanceBass</span>(lr_lev, bass_lev, hpf_bypass, cutoff);</p>
|
|
<p class=desc>Configures the bass enhancement by setting the levels of the
|
|
original stereo signal and the bass-enhanced mono level which will be mixed together.
|
|
The high-pass filter may be enabled (0) or bypassed (1). The cutoff frequency is specified
|
|
as follows:
|
|
</p>
|
|
<pre class="desc">
|
|
value frequency
|
|
0 80Hz
|
|
1 100Hz
|
|
2 125Hz
|
|
3 150Hz
|
|
4 175Hz
|
|
5 200Hz
|
|
6 225Hz
|
|
</pre>
|
|
<p class=desc>When changing bass level, call this function repeatedly to ramp up or down the bass in
|
|
steps of 0.5dB, to avoid pops.
|
|
</p>
|
|
|
|
<p class=func><span class=keyword>eqSelect</span>(n);</p>
|
|
<p class=desc>Selects the type of frequency control, where <em>n</em> is
|
|
one of</p>
|
|
<p class=desc><b>FLAT_FREQUENCY (0)</b><br>
|
|
Equalizers and tone controls disabled, flat frequency response.</p>
|
|
<p class=desc><b>PARAMETRIC_EQUALIZER (1)</b><br>
|
|
Enables the 7-band parametric equalizer, thus disabling the
|
|
tone controls and graphic equalizer.</p>
|
|
<p class=desc><b>TONE_CONTROLS (2)</b><br>
|
|
Enables bass and treble tone controls, disabling the parametric
|
|
equalization and graphic equalizer.</p>
|
|
<p class=desc><b>GRAPHIC_EQUALIZER (3)</b><br>
|
|
Enables the five-band graphic equalizer, disabling the parametric
|
|
equalization and tone controls.</p>
|
|
|
|
|
|
<p class=func><span class=keyword>eqBands</span>(bass, treble);</p>
|
|
<p class=desc>Configures bass and treble tone controls, which are
|
|
implemented as one second order low pass filter (bass) in parallel with
|
|
one second order high pass filter (treble).
|
|
</p>
|
|
<p class=desc>When changing bass or treble level, call this function repeatedly to ramp
|
|
up or down the level in steps of 0.04 (=0.5dB) or so, to avoid pops.
|
|
</p>
|
|
<p class=func><span class=keyword>eqBands</span>(bass, mid_bass, midrange, mid_treble, treble);</p>
|
|
<p class=desc>Configures the graphic equalizer. It is implemented by five parallel,
|
|
second order biquad filters with fixed frequencies of 115Hz, 330Hz, 990Hz, 3kHz,
|
|
and 9.9kHz. Each band has a range of adjustment from 1.00 (+12dB) to -1.00 (-11.75dB).
|
|
</p>
|
|
<p class=func><span class=keyword>eqBand</span>(bandNum, n);</p>
|
|
<p class=desc>Configures the gain or cut on one band in the graphic equalizer.
|
|
<em>bandnum</em> can range from 1 to 5; <em>n</em> is a float in the range 1.00 to -1.00.
|
|
</p>
|
|
<p class=desc>When changing a band, call this function repeatedly to ramp up the gain in steps of 0.5dB,
|
|
to avoid pops.
|
|
</p>
|
|
|
|
<p class=func><span class=keyword>eqFilter</span>(filterNum, filterParameters);</p>
|
|
<p class=desc>Configurs the parametric equalizer. The number of filters (1 to 7)
|
|
is specified along with a pointer to an array of filter coefficients.
|
|
The parametric equalizer is implemented using 7 cascaded, second order bi-quad
|
|
filters whose frequencies, gain, and Q may be freely configured, but each filter
|
|
can only be specified as a set of filter coefficients.
|
|
</p>
|
|
<p class=func><span class=keyword>eqFilterCount</span>(n);</p>
|
|
<p class=desc>Enables zero or more of the already enabled parametric filters.
|
|
</p>
|
|
|
|
<h3>Examples</h3>
|
|
<p>Nearly all of the Teensy Audio and OpenAudio_ArduinoLibrary (F32 floating point)
|
|
library's examples use this object. These examples
|
|
for the Teensy Audio Library demonstrate its special features.
|
|
</p>
|
|
<p class=exam>File > Examples > Audio > HardwareTesting > PassThroughStereo
|
|
</p>
|
|
<p class=exam>File > Examples > Audio > HardwareTesting > SGTL5000 > dap_bass_enhance
|
|
</p>
|
|
<p class=exam>File > Examples > Audio > HardwareTesting > SGTL5000 > dap_avc_agc
|
|
</p>
|
|
<p class=exam>File > Examples > Audio > HardwareTesting > SGTL5000 > balanceDAC
|
|
</p>
|
|
<p class=exam>File > Examples > Audio > HardwareTesting > SGTL5000 > balanceHP
|
|
</p>
|
|
<p class=exam>File > Examples > Audio > HardwareTesting > SGTL5000 > CalcBiquadToneControlDAP
|
|
</p>
|
|
<p class=exam>File > Examples > Audio > HardwareTesting > SGTL5000 > VolumeRamp
|
|
</p>
|
|
|
|
<h3>Notes</h3>
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioControlSGTL5000">
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
<script type="text/x-red" data-help-name="AudioControlWM8731">
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>Control a WM8731 chip in slave mode, where it receives all clocks from Teensy</p>
|
|
</div>
|
|
<h3>Audio Connections</h3>
|
|
<p>This object has no audio inputs or outputs. Separate i2s objects
|
|
are used to send and receive audio data. I2S master mode objects
|
|
must be used, since this control object configures the WM8731 into
|
|
slave mode.
|
|
</p>
|
|
<h3>Functions</h3>
|
|
<p class=func><span class=keyword>enable</span>();</p>
|
|
<p class=desc>Turn on the WS8731.
|
|
</p>
|
|
<p class=func><span class=keyword>disable</span>();</p>
|
|
<p class=desc>not implemented
|
|
</p>
|
|
<p class=func><span class=keyword>volume</span>(level);</p>
|
|
<p class=desc>Set the headphone volume level. Range is 0 to 1.0.
|
|
</p>
|
|
<p class=func><span class=keyword>inputLevel</span>(level);</p>
|
|
<p class=desc>Adjust the line level input gain. Range is 0 to 1.0.
|
|
</p>
|
|
<p class=func><span class=keyword>inputSelect</span>(input);</p>
|
|
<p class=desc>Select which input to use: AUDIO_INPUT_LINEIN or AUDIO_INPUT_MIC.
|
|
</p>
|
|
<!--
|
|
<h3>Examples</h3>
|
|
<p class=exam>File > Examples > Audio >
|
|
</p>
|
|
-->
|
|
<h3>Notes</h3>
|
|
<p></p>
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioControlWM8731">
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
<script type="text/x-red" data-help-name="AudioControlWM8731master">
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>Control a WM8731 chip in master mode, where it controls all I2S timing.</p>
|
|
</div>
|
|
<h3>Audio Connections</h3>
|
|
<p>This object has no audio inputs or outputs. Separate i2s objects
|
|
are used to send and receive audio data. I2S slave mode objects
|
|
must be used, since this control object configures the WM8731 into
|
|
master mode.
|
|
</p>
|
|
<h3>Functions</h3>
|
|
<p class=func><span class=keyword>enable</span>();</p>
|
|
<p class=desc>Turn on the WS8731, in I2S Master mode. I2S slave mode
|
|
communication must be used by Teensy.
|
|
</p>
|
|
<p class=func><span class=keyword>disable</span>();</p>
|
|
<p class=desc>not implemented
|
|
</p>
|
|
<p class=func><span class=keyword>volume</span>(level);</p>
|
|
<p class=desc>Set the headphone volume level. Range is 0 to 1.0.
|
|
</p>
|
|
<p class=func><span class=keyword>inputLevel</span>(level);</p>
|
|
<p class=desc>Adjust the line level input gain. Range is 0 to 1.0.
|
|
</p>
|
|
<p class=func><span class=keyword>inputSelect</span>(input);</p>
|
|
<p class=desc>Select which input to use: AUDIO_INPUT_LINEIN or AUDIO_INPUT_MIC.
|
|
</p>
|
|
<h3>Examples</h3>
|
|
<p class=exam>File > Examples > Audio > HardwareTesting > WM8731MikroSine
|
|
</p>
|
|
<h3>Notes</h3>
|
|
<p>The WM8731 will implement a sample rate of its crystal frequency divided by 256.
|
|
To get the 44.1 kHz sample rate the Teensy Audio Library expects, an
|
|
11.2896 MHz crystal should be used.
|
|
</p>
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioControlWM8731master">
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
<script type="text/x-red" data-help-name="AudioControlAK4558">
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>Control the AK4558 chip on the <a href="https://hackaday.io/project/8567-hifi-audio-codec-module" target="_blank">HiFi Audio CODEC Module</a>
|
|
in slave mode, where the Teensy controls all I2S timing.</p>
|
|
</div>
|
|
<h3>Audio Connections</h3>
|
|
<p>This object has no audio inputs or outputs. Separate I2S objects
|
|
are used to send and receive audio data.
|
|
</p>
|
|
<h3>Functions</h3>
|
|
<p class=func><span class=keyword>enable</span>();</p>
|
|
<p class=desc>Enables the CODEC to work with 44.1 KHz - 16 bit data. This function does not enable the ADC/DAC modules.
|
|
</p>
|
|
<p class=func><span class=keyword>enableIn</span>();</p>
|
|
<p class=desc>Enables the ADC module.
|
|
</p>
|
|
<p class=func><span class=keyword>enableOut</span>();</p>
|
|
<p class=desc>Enables the DAC module.
|
|
</p>
|
|
<p class=func><span class=keyword>disable</span>();</p>
|
|
<p class=desc>Disables the ADC and the DAC modules.
|
|
</p>
|
|
<p class=func><span class=keyword>disableIn</span>();</p>
|
|
<p class=desc>Disable the ADC module.
|
|
</p>
|
|
<p class=func><span class=keyword>disableOut</span>();</p>
|
|
<p class=desc>Disable the DAC module.
|
|
</p>
|
|
<p class=func><span class=keyword>volume</span>(level);</p>
|
|
<p class=desc>Accepts a float in range 0.0-1.0 and sets the line output volume accordingly.
|
|
</p>
|
|
<p class=func><span class=keyword>volumeLeft</span>(level);</p>
|
|
<p class=desc>Accepts a float in range 0.0-1.0 and sets the left line output volume accordingly.
|
|
</p>
|
|
<p class=func><span class=keyword>volumeRight</span>(level);</p>
|
|
<p class=desc>Accepts a float in range 0.0-1.0 and sets the right line output volume accordingly.
|
|
</p>
|
|
<p class=func><span class=keyword>inputLevel</span>(level);</p>
|
|
<p class=desc>NOT SUPPORTED BY THE AK4558
|
|
</p>
|
|
<p class=func><span class=keyword>inputSelect</span>(input);</p>
|
|
<p class=desc>not implemented yet
|
|
</p>
|
|
<h3>Examples</h3>
|
|
<p class=exam>File > Examples > Audio > HardwareTesting > AK4558 > PassthroughTest
|
|
</p>
|
|
<p class=exam>File > Examples > Audio > HardwareTesting > AK4558 > SineOutTest
|
|
</p>
|
|
<h3>Notes</h3>
|
|
<p>TODO: Implement inputSelect() function to enable mono left, mono right, stereo operation.</p>
|
|
<p>TODO: Implement ADC and DAC filters control.</p>
|
|
<p>TODO: Implement DAC level attenuator attack rate modifier.</p>
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioControlAK4558">
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
<script type="text/x-red" data-help-name="AudioControlCS4272">
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>Control the CS4272 chip on the <a href="https://hackaday.io/project/5912-teensy-super-audio-board" target="_blank">Super Audio Board</a>.
|
|
</p>
|
|
<p>TODO: does this control object put the CS4272 into I2S master or slave mode</p>
|
|
</div>
|
|
<h3>Audio Connections</h3>
|
|
<p>This object has no audio inputs or outputs. Separate I2S objects
|
|
are used to send and receive audio data.
|
|
</p>
|
|
<h3>Functions</h3>
|
|
<p class=func><span class=keyword>enable</span>();</p>
|
|
<p class=desc>Enables the CODEC to work with 44.1 KHz - 16 bit data. This function does not enable the ADC/DAC modules.
|
|
</p>
|
|
<p class=func><span class=keyword>volume</span>(vol);</p>
|
|
<p class=desc>Set the volume level. Range is 0 to 1.0.
|
|
</p>
|
|
<p class=func><span class=keyword>volume</span>(left, right);</p>
|
|
<p class=desc>Set the volume level. Range is 0 to 1.0.
|
|
</p>
|
|
<p class=func><span class=keyword>dacVolume</span>(vol);</p>
|
|
<p class=desc>Set the volume level. Range is 0 to 1.0. TODO: what's the
|
|
distinction between volume() and dacVolume()?
|
|
</p>
|
|
<p class=func><span class=keyword>dacVolume</span>(left, right);</p>
|
|
<p class=desc>Set the volume level. Range is 0 to 1.0.
|
|
</p>
|
|
|
|
<p class=func><span class=keyword>muteOutput</span>();</p>
|
|
<p class=desc>TODO: description
|
|
</p>
|
|
<p class=func><span class=keyword>unmuteOutput</span>();</p>
|
|
<p class=desc>TODO: description
|
|
</p>
|
|
<p class=func><span class=keyword>muteInput</span>();</p>
|
|
<p class=desc>TODO: description
|
|
</p>
|
|
<p class=func><span class=keyword>unmuteInput</span>();</p>
|
|
<p class=desc>TODO: description
|
|
</p>
|
|
<p class=func><span class=keyword>enableDither</span>();</p>
|
|
<p class=desc>TODO: description
|
|
</p>
|
|
<p class=func><span class=keyword>disableDither</span>();</p>
|
|
<p class=desc>TODO: description
|
|
</p>
|
|
|
|
<h3>Hardware</h3>
|
|
<p>Pin 2 must be connected to the CS4272 reset. SDA & SCL are used for all control.
|
|
</p>
|
|
|
|
<h3>Notes</h3>
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioControlCS4272">
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
<script type="text/x-red" data-help-name="AudioControlCS42448">
|
|
<h3>Summary</h3>
|
|
<div class=tooltipinfo>
|
|
<p>Control the CS42448 chip in TDM mode, for 6 inputs and 8 outputs.
|
|
</p>
|
|
<p align=center><img src="img/cs42448.jpg"></p>
|
|
</div>
|
|
</div>
|
|
<h3>Audio Connections</h3>
|
|
<p>This object has no audio inputs or outputs. Separate TDM objects
|
|
are used to send and receive audio data.
|
|
</p>
|
|
<h3>Functions</h3>
|
|
<p class=func><span class=keyword>enable</span>();</p>
|
|
<p class=desc>Enables the CS42448 to work in TDM mode.
|
|
</p>
|
|
<p class=func><span class=keyword>volume</span>(level);</p>
|
|
<p class=desc>Set the volume level for all output channels. Range is 0 to 1.0.
|
|
</p>
|
|
<p class=func><span class=keyword>volume</span>(channel, level);</p>
|
|
<p class=desc>Set the volume level for a single output. Channel is 1 to 8. Range is 0 to 1.0.
|
|
</p>
|
|
<p class=func><span class=keyword>inputLevel</span>(level);</p>
|
|
<p class=desc>Set the input gain level for all input channels. Range is 0 to 15.85.
|
|
</p>
|
|
<p class=func><span class=keyword>inputLevel</span>(channel, level);</p>
|
|
<p class=desc>Set the input gain level for a single input. Channel is 1 to 6. Range is 0 to 15.85.
|
|
</p>
|
|
<h3>Hardware</h3>
|
|
<p>Tested with this <a href="https://oshpark.com/shared_projects/2Yj6rFaW">
|
|
CS42448 Circuit Board</a>.
|
|
</p>
|
|
<p align=center><img src="img/tdm.jpg"></p>
|
|
</div>
|
|
<h3>Notes</h3>
|
|
</script>
|
|
<script type="text/x-red" data-template-name="AudioControlCS42448">
|
|
<div class="form-row">
|
|
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
|
|
<input type="text" id="node-input-name" placeholder="Name">
|
|
</div>
|
|
</script>
|
|
|
|
|
|
</body>
|
|
</html>
|
|
|