screen.js, RTCMultiConnection-v1.8, part-fo-screen demo, updated.

muaz-khan · Jun 6, 2014 · 055a4b9 · 055a4b9
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diff --git a/RTCMultiConnection/Library/RTCMultiConnection-v1.8.js b/RTCMultiConnection/Library/RTCMultiConnection-v1.8.js
diff --git a/RTCMultiConnection/RTCMultiConnection-v1.2-and-earlier.md b/RTCMultiConnection/RTCMultiConnection-v1.2-and-earlier.md
diff --git a/RTCMultiConnection/RTCMultiConnection-v1.3.md b/RTCMultiConnection/RTCMultiConnection-v1.3.md
diff --git a/RTCMultiConnection/changes-log.md b/RTCMultiConnection/changes-log.md
diff --git a/RTCMultiConnection/changes-log/index.html b/RTCMultiConnection/changes-log/index.html
@@ -74,6 +74,21 @@ <h2 id="v1.8"><a href="http://www.rtcmulticonnection.org/changes-log/#v1.8">v1.8
 
                 <ol>
 
+                    <li>
+                        <code>captureUserMediaOnDemand</code> added for <code>connection.open</code> method:
+                        <pre class="sh_javascript">
+// it is enabled by default
+// it means that "getUserMedia" API for initiator will 
+// be invoked only when required.
+// i.e. when first participant is detected.
+
+// you can disable it by setting it to "false"
+connection.open({
+    captureUserMediaOnDemand: false
+});
+</pre>
+                    </li>
+
                     <li>
                         <code>connection.DetectRTC.screen.getChromeExtensionStatus</code> added.
                         <pre class="sh_javascript">

diff --git a/RTCMultiConnection/demos/remote-stream-forwarding.html b/RTCMultiConnection/demos/remote-stream-forwarding.html
@@ -14,6 +14,7 @@
         <link rel="author" type="text/html" href="https://plus.google.com/+MuazKhan">
         <meta name="author" content="Muaz Khan">
         <meta http-equiv="X-UA-Compatible" content="IE=edge,chrome=1">
+
         <link rel="stylesheet" href="https://www.webrtc-experiment.com/style.css">
 
         <style>
@@ -58,7 +59,6 @@
             document.createElement('footer');
         </script>
 
-        <!-- scripts used for video-conferencing -->
         <script src="https://www.webrtc-experiment.com/RTCMultiConnection-v1.8.js"> </script>
         <script src="https://www.webrtc-experiment.com/firebase.js"> </script>
     </head>
@@ -91,6 +91,10 @@ <h1>
             </header>
 
             <div class="github-stargazers"></div>
+
+            <blockquote class="inner" style="text-align: center;">
+                Chrome is <a href="https://code.google.com/p/webrtc/issues/detail?id=2192#c15">still not supporting</a> remote audio forwarding. It will forward only remote <strong>video</strong> stream.
+            </blockquote>
 
             <section class="experiment">
                 <div id="videos-container"></div>
@@ -160,7 +164,12 @@ <h2>How to use?</h2>
 
                     // http://www.rtcmulticonnection.org/docs/onstream/
                     mainConnection.onstream = function(e) {
-                        dummyConnection.attachStreams.push(e.stream /* .clone() */ );
+                        if(mainConnection.UA.Chrome) {
+                            // chrome doesn't yet permits remote audio forwarding
+                            // https://code.google.com/p/webrtc/issues/detail?id=2192#c15
+                            var stream = new webkitMediaStream( e.stream.getVideoTracks() );
+                            dummyConnection.attachStreams.push(stream);
+                        }
                         videosContainer.appendChild(e.mediaElement);
                         document.querySelector('#forward-main-session').disabled = false;
                     };
@@ -196,7 +205,9 @@ <h2>How to use?</h2>
                         this.disabled = true;
 
                         // http://www.rtcmulticonnection.org/docs/open/
-                        dummyConnection.open();
+                        dummyConnection.open({
+                            captureUserMediaOnDemand: false
+                        });
                     };
                 </script>
             </section>