Merge dev into master

AUTHORS

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+The WebRTC Project Authors
+The Chromium Authors
+Temasys Communications

CONTRIBUTING.md

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+WebRTC welcomes patches/pulls for features and bug fixes.
+
+For contributors external to Google, follow the instructions given in the [Google Individual Contributor License Agreement](https://cla.developers.google.com/about/google-individual).
+
+In all cases, contributors must sign a contributor license agreement before a contribution can be accepted. Please complete the agreement for an [individual](https://developers.google.com/open-source/cla/individual) or a [corporation](https://developers.google.com/open-source/cla/corporate) as appropriate.
+
+If you plan to add a significant component or large chunk of code, we recommend you bring this up on the [webrtc-discuss group](https://groups.google.com/forum/#!forum/discuss-webrtc) for a design discussion before writing code.
+
+If appropriate, write a unit test which demonstrates that your code functions as expected. Tests are the best way to ensure that future contributors do not break your code accidentally.
+
+To request a change or addition, you must [submit a pull request](https://help.github.com/categories/collaborating/).
+
+If you need to update the adapter.js version, run "npm install" and submit a CL with the resulting change - this will update the version in src/js/adapter.js.
+
+WebRTC developers monitor outstanding pull requests. They may request changes to the pull request before accepting. They will also verify that a CLA has been signed.
+
+The [Developer's Guide](https://bit.ly/webrtcdevguide) for this repo has more detailed information about code style, structure and validation.
+
+The [README](test/README.md) in the test folder has all the information needed for setting up the development enviroment and how create tests for the demo pages.

Gruntfile.js

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+'use strict';
+
+/* globals module */
+
+module.exports = function(grunt) {
+  // configure project
+  grunt.initConfig({
+    // make node configurations available
+    pkg: grunt.file.readJSON('package.json'),
+    csslint: {
+      options: {
+        csslintrc: '.csslintrc'
+      },
+      strict: {
+        options: {
+          import: 2
+        },
+        src: ['src/content/**/*.css', '!src/content/**/*_nolint.css']
+      },
+      lax: {
+        options: {
+          import: false
+        },
+        src: ['src/content/**/*.css', '!src/content/**/*_nolint.css']
+      }
+    },
+    eslint: {
+      options: {
+        configFile: '.eslintrc'
+      },
+      target: ['src/content/**/*.js', 'test/*.js']
+    },
+    githooks: {
+      all: {
+        'pre-commit': 'csslint htmlhint eslint'
+      }
+    },
+    htmlhint: {
+      html1: {
+        src: [
+          'src/content/datachannel/**/index.html',
+          'src/content/getusermedia/**/index.html',
+          'src/content/peerconnection/**/index.html'
+        ]
+      }
+    },
+    // Leaving this as a manual step as the extension is not updated regularly.
+    compress: {
+      main: {
+        options: {
+          mode: 'zip',
+          archive: 'release/desktopCaptureExtension.zip'
+        },
+        files: [
+          {
+            expand: true,
+            cwd: 'src/content/extensions/desktopcapture/extension',
+            src: '**',
+            dest: 'desktopCaptureExtension',
+            isfile: true
+          }
+        ]
+      }
+    }
+  });
+
+  // enable plugins
+  grunt.loadNpmTasks('grunt-contrib-compress');
+  grunt.loadNpmTasks('grunt-contrib-csslint');
+  grunt.loadNpmTasks('grunt-eslint');
+  grunt.loadNpmTasks('grunt-githooks');
+  grunt.loadNpmTasks('grunt-htmlhint');
+
+  // set default tasks to run when grunt is called without parameters
+  grunt.registerTask('default', ['csslint', 'htmlhint', 'eslint']);
+  // also possible to call JavaScript directly in registerTask()
+  // or to call external tasks with grunt.loadTasks()
+};

ISSUE_TEMPLATE.md

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+**Browsers and versions affected**
+
+
+**Description**
+
+
+**Steps to reproduce**
+
+
+**Expected results**
+
+
+**Actual results**

PULL_REQUEST_TEMPLATE.md

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+**Description**
+
+
+**Purpose**

README.md

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 # WebRTC code samples #
 
-This repository hosts forks of the different [samples provided by Google](https://github.com/GoogleChrome/webrtc) with the  few changes needed so they would work on IE and Safari with the [plugin provided by Temasys](http://plugin.temasys.com.sg/)
+This is a repository for the WebRTC Javascript code samples.
 
-All of the samples use [adapter.js](https://github.com/Temasys/AdapterJS), a shim to insulate apps from spec changes and prefix differences. In fact, the standards and protocols used for WebRTC implementations are highly stable, and there are only a few prefixed names. 
+It is originally a fork of [https://github.com/webrtc/samples](https://github.com/webrtc/samples), but updated to integrate the [Temasys Plugin](plugin.temasys.com.sg), and work on Internet Explorer and Safari.
 
-NB for chrome: all samples that use `getUserMedia()` must be run from a server. Calling `getUserMedia()` from a file:// URL will result in a PERMISSION_DENIED NavigatorUserMediaError.  See [What are some chromium command-line flags relevant to WebRTC development/testing?](http://www.webrtc.org/chrome#TOC-What-are-some-chromium-command-line-flags-relevant-to-WebRTC-development-testing-) for relevant flags.
+Some of the samples use new browser features. They may only work in Chrome Canary, Firefox Beta, Microsoft Edge (available with Windows 10), and latest versions of the [Temasys Plugin](plugin.temasys.com.sg), and may require flags to be set.
 
-Patches and issues welcome!
+All of the samples use [AdapterJS](https://github.com/Temasys/AdapterJS), a shim to insulate apps from spec changes and prefix differences. In fact, the standards and protocols used for WebRTC implementations are highly stable, and there are only a few prefixed names. For full interop information, see [webrtc.org/web-apis/interop](http://www.webrtc.org/web-apis/interop).
 
-Tests updated today
-=========
+In Chrome and Opera, all samples that use `getUserMedia()` must be run from a server. Calling `getUserMedia()` from a file:// URL will work in Firefox and the Temasys Plugin, but fail silently in Chrome and Opera.
 
-[getUserMedia()](https://github.com/TemasysCommunications/Google-WebRTC-Samples/tree/master/samples/web/content/getusermedia/gum) 
+[webrtc.org/testing](http://www.webrtc.org/testing) lists command line flags useful for development and testing with Chrome.
 
-[getUserMedia with camera/mic selection](https://github.com/TemasysCommunications/Google-WebRTC-Samples/tree/master/samples/web/content/getusermedia/source)
+For more information about WebRTC, we maintain a list of [WebRTC Resources](https://docs.google.com/document/d/1idl_NYQhllFEFqkGQOLv8KBK8M3EVzyvxnKkHl4SuM8/edit). If you've never worked with WebRTC, we recommend you start with the 2013 Google I/O [WebRTC presentation](http://www.youtube.com/watch?v=p2HzZkd2A40).
 
-[getUserMedia in iFrame](https://github.com/TemasysCommunications/Google-WebRTC-Samples/tree/master/samples/web/content/getusermedia/iframe)
+Patches and issues welcome! See [CONTRIBUTING](https://github.com/Temasys/Google-WebRTC-Samples/blob/dev/CONTRIBUTING.md) for instructions. All contributors must sign a contributor license agreement before code can be accepted. Please complete the agreement for an [individual](https://developers.google.com/open-source/cla/individual) or a [corporation](https://developers.google.com/open-source/cla/corporate) as appropriate.
+The [Developer's Guide](https://bit.ly/webrtcdevguide) for this repo has more information about code style, structure and validation.
+Head over to [test/README.md](https://github.com/Temasys/Google-WebRTC-Samples/blob/dev/test/README.md) and get started developing.
 
-[getUserMedia in iFrame](https://github.com/TemasysCommunications/Google-WebRTC-Samples/tree/master/samples/web/content/getusermedia/playPause)
+## The demos ##
 
-[ICE candidate gathering](https://github.com/TemasysCommunications/Google-WebRTC-Samples/tree/master/samples/web/content/peerconnection/trickle-ice)
+### getUserMedia ###
 
-[Peer Connection](https://github.com/TemasysCommunications/Google-WebRTC-Samples/tree/master/samples/web/content/peerconnection/pc1)
+[Basic getUserMedia demo](https://github.com/Temasys/Google-WebRTC-Samples/src/content/getusermedia/gum/)
 
-[Peer Connection States](https://github.com/TemasysCommunications/Google-WebRTC-Samples/tree/master/samples/web/content/peerconnection/states)
+<!-- [getUserMedia + canvas](https://github.com/Temasys/Google-WebRTC-Samples/src/content/getusermedia/canvas/) -->
 
-[Audio-only peer connection](https://github.com/TemasysCommunications/Google-WebRTC-Samples/tree/master/samples/web/content/peerconnection/audio)
+<!-- [getUserMedia + canvas + CSS Filters](https://github.com/Temasys/Google-WebRTC-Samples/src/content/getusermedia/filter/) -->
 
-[Multiple peer connections](https://github.com/TemasysCommunications/Google-WebRTC-Samples/tree/master/samples/web/content/peerconnection/multiple)
+[getUserMedia with resolution constraints](https://github.com/Temasys/Google-WebRTC-Samples/src/content/getusermedia/resolution/)
 
-[Multiple relays](https://github.com/TemasysCommunications/Google-WebRTC-Samples/tree/master/samples/web/content/peerconnection/multiple-relay)
+[getUserMedia with camera, mic and speaker selection](https://github.com/Temasys/Google-WebRTC-Samples/src/content/getusermedia/source/)
 
-[Munge SDP](https://github.com/TemasysCommunications/Google-WebRTC-Samples/tree/master/samples/web/content/peerconnection/munge-sdp)
+[Audio-only getUserMedia output to local audio element](https://github.com/Temasys/Google-WebRTC-Samples/src/content/getusermedia/audio/)
 
-[Data channels](https://github.com/TemasysCommunications/Google-WebRTC-Samples/tree/master/samples/web/content/datachannel)
+<!-- [Audio-only getUserMedia displaying volume](https://github.com/Temasys/Google-WebRTC-Samples/src/content/getusermedia/volume/) -->
 
-[Data channels with arraybuffers](https://github.com/TemasysCommunications/Google-WebRTC-Samples/tree/master/samples/web/content/datachannel-arraybuffer)
+<!-- [Face tracking](https://github.com/Temasys/Google-WebRTC-Samples/src/content/getusermedia/face/) -->
 
-[AppRTC](https://github.com/TemasysCommunications/Google-WebRTC-Samples/tree/master/samples/web/content/apprtc)
+<!-- [Record stream](https://github.com/Temasys/Google-WebRTC-Samples/src/content/getusermedia/record/) -->
 
-Tests to be updated in the future
-=========
+[getUserMedia in an iFrame](https://github.com/Temasys/Google-WebRTC-Samples/src/content/getusermedia/iframe)
 
-[getUserMedia + canvas](https://googlechrome.github.io/webrtc/samples/web/content/getusermedia/canvas)
+[Screensharing]](https://github.com/Temasys/Google-WebRTC-Samples/src/content/getusermedia/screensharing)
 
-[getUserMedia + canvas + CSS Filters](https://googlechrome.github.io/webrtc/samples/web/content/getusermedia/filter)
+### Devices ###
 
-[getUserMedia with resolution constraints](https://googlechrome.github.io/webrtc/samples/web/content/getusermedia/resolution)
+[Select camera, microphone and speaker](https://github.com/Temasys/Google-WebRTC-Samples/src/content/devices/input-output/)
 
-[Audio-only getUserMedia output to local audio element](https://googlechrome.github.io/webrtc/samples/web/content/getusermedia/audio)
+[Select media source and audio output](https://github.com/Temasys/Google-WebRTC-Samples/src/content/devices/multi/)
 
-[Audio-only getUserMedia displaying volume](https://googlechrome.github.io/webrtc/samples/web/content/getusermedia/volume)
+### RTCPeerConnection ###
 
-[Face tracking](https://googlechrome.github.io/webrtc/samples/web/content/getusermedia/face)
+[Basic peer connection](https://github.com/Temasys/Google-WebRTC-Samples/src/content/peerconnection/pc1/)
 
-[Accept incoming peer connection](http://googlechrome.github.io/webrtc/samples/web/content/pr-answer)
+[Audio-only peer connection](https://github.com/Temasys/Google-WebRTC-Samples/src/content/peerconnection/audio/)
 
-[Use pranswer when setting up a peer connection](https://googlechrome.github.io/webrtc/samples/web/content/peerconnection/pr-answer)
+[Multiple peer connections at once](https://github.com/Temasys/Google-WebRTC-Samples/src/content/peerconnection/multiple/)
 
-[Adjust constraints, view stats](https://googlechrome.github.io/webrtc/samples/web/content/peerconnection/constraints)
+[Forward output of one peer connection into another](https://github.com/Temasys/Google-WebRTC-Samples/src/content/peerconnection/multiple-relay/)
 
-[Display createOffer output](https://googlechrome.github.io/webrtc/samples/web/content/peerconnection/create-offer)
+[Munge SDP parameters](https://github.com/Temasys/Google-WebRTC-Samples/src/content/peerconnection/munge-sdp/)
 
-[Use RTCDTMFSender](https://googlechrome.github.io/webrtc/samples/web/content/peerconnection/dtmf)
+<!-- [Use pranswer when setting up a peer connection](https://github.com/Temasys/Google-WebRTC-Samples/src/content/peerconnection/pr-answer/) -->
 
-[Web Audio output as input to peer connection](https://googlechrome.github.io/webrtc/samples/web/content/peerconnection/webaudio-input)
+[Adjust constraints, view stats](https://github.com/Temasys/Google-WebRTC-Samples/src/content/peerconnection/constraints/)
 
+<!-- [Display createOffer output](https://github.com/Temasys/Google-WebRTC-Samples/src/content/peerconnection/create-offer/) -->
 
+[Use RTCDTMFSender](https://github.com/Temasys/Google-WebRTC-Samples/src/content/peerconnection/dtmf/)
 
+[Display peer connection states](https://github.com/Temasys/Google-WebRTC-Samples/src/content/peerconnection/states/)
+
+[ICE candidate gathering from STUN/TURN servers](https://github.com/Temasys/Google-WebRTC-Samples/src/content/peerconnection/trickle-ice/)
+
+[Do an ICE restart](https://github.com/Temasys/Google-WebRTC-Samples/src/content/peerconnection/restart-ice/)
+
+<!-- [Web Audio output as input to peer connection](https://github.com/Temasys/Google-WebRTC-Samples/src/content/peerconnection/webaudio-input/) -->
+
+<!-- [Peer connection as input to Web Audio](https://github.com/Temasys/Google-WebRTC-Samples/src/content/peerconnection/webaudio-output/) -->
+
+### RTCDataChannel ###
+
+[Transmit text](https://github.com/Temasys/Google-WebRTC-Samples/src/content/datachannel/basic/)
+
+<!-- [Transfer a file](https://github.com/Temasys/Google-WebRTC-Samples/src/content/datachannel/filetransfer/) -->
+
+<!-- [Transfer data](https://github.com/Temasys/Google-WebRTC-Samples/src/content/datachannel/datatransfer/) -->
+
+[ArrayBuffer sending](https://github.com/Temasys/Google-WebRTC-Samples/src/content/datachannel/arraybuffer/)
+
+<!-- ### Video chat ###
+
+[AppRTC video chat client](https://apprtc.appspot.com/) powered by Google App Engine
+
+[AppRTC URL parameters](https://apprtc.appspot.com/params.html)
+ -->

google1b7eb21c5b594ba0.html

@@ -0,0 +1 @@
+google-site-verification: google1b7eb21c5b594ba0.html