Adding WebRTC-in-Chrome test case that includes an SDP offer with B=AS:xx

content/test/data/media/peerconnection-call.html

Index: content/test/data/media/peerconnection-call.html
diff --git a/content/test/data/media/peerconnection-call.html b/content/test/data/media/peerconnection-call.html
index b0034149ad1a10e23730103ae8db33f4fd0b557a..bb9f6ab92303b740e850a4d9327a4b28254eaf80 100644
--- a/content/test/data/media/peerconnection-call.html
+++ b/content/test/data/media/peerconnection-call.html
@@ -152,6 +152,19 @@
     };
   }
 
+  // Test that we can negotiate a call with an SDP offer that includes a
+  // b=AS:XX line to control audio and video bandwidth
+  function negotiateOfferWithBLine(force_isac_16k) {
+    if (force_isac_16k)
+      transformSdp = forceIsac16k_;
+    createConnections(null);
+    transformSdp = addBandwithControl;

[wjia(left Chromium), 2013/12/13 18:24:19]

Shall you call forceIsac16k_ inside addBandwithControl? I am not sure about the
calling sequence on transformSdp.

+    navigator.webkitGetUserMedia({audio: true, video: true},
+        addStreamToBothConnectionsAndNegotiate, printGetUserMediaError);
+    waitForVideo('remote-view-1');
+    waitForVideo('remote-view-2');
+  }
+
   // Test that we can setup call with legacy settings.
   function callWithLegacySdp() {
     transformSdp = function(sdp) {
@@ -643,6 +656,14 @@
     return offerSdp;
   }
 
+  function addBandwithControl(offerSdp) {
+    offerSdp = offerSdp.replace('a=mid:audio\r\n', 'a=mid:audio\r\n'+
+                                'b=AS:16\r\n');
+    offerSdp = offerSdp.replace('a=mid:video\r\n', 'a=mid:video\r\n'+
+                                'b=AS:512\r\n');
+    return offerSdp;
+  }
+
   function removeBundle(sdp) {
     return sdp.replace(/a=group:BUNDLE .*\r\n/g, '');
   }