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

content/browser/media/webrtc_browsertest.cc

Index: content/browser/media/webrtc_browsertest.cc
diff --git a/content/browser/media/webrtc_browsertest.cc b/content/browser/media/webrtc_browsertest.cc
index 47847a7d53bac2b7b1776d4602123880539f273f..cca195b8c4aee5d93b967a054168fd1be9a4462b 100644
--- a/content/browser/media/webrtc_browsertest.cc
+++ b/content/browser/media/webrtc_browsertest.cc
@@ -360,6 +360,18 @@ IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, NegotiateNonCryptoCall) {
   ExpectTitle("OK");
 }
 
+// This test can negotiate an SDP offer that includes a b=AS:xx to control
+// the bandwidth for audio and video
+IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, NegotiateOfferWithBLine) {
+  ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady());
+
+  GURL url(embedded_test_server()->GetURL("/media/peerconnection-call.html"));
+  NavigateToURL(shell(), url);
+
+  EXPECT_TRUE(ExecuteJavascript("negotiateOfferWithBLine();"));

[wjia(left Chromium), 2013/12/12 22:04:35]

It seems this test enables both audio and video. On Android, we need to force
using ISAC/16000, just like EstablishAudioVideoCallAndMeasureOutputLevel below.
Is it needed here, too?

+  ExpectTitle("OK");
+}
+
 // This test will make a complete PeerConnection-based call using legacy SDP
 // settings: GIce, external SDES, and no BUNDLE.
 #if defined(OS_WIN) && defined(USE_AURA)