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

content/browser/media/webrtc_browsertest.cc

 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "base/command_line.h"
 #include "base/debug/trace_event_impl.h"
 #include "base/json/json_reader.h"
 #include "base/strings/stringprintf.h"
 #include "base/strings/utf_string_conversions.h"
 #include "base/values.h"
@@ skipping 342 matching lines @@
 IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, NegotiateNonCryptoCall) {
   ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady());
 
   GURL url(embedded_test_server()->GetURL("/media/peerconnection-call.html"));
   NavigateToURL(shell(), url);
 
   EXPECT_TRUE(ExecuteJavascript("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)
 // Disabled for win7_aura, see http://crbug.com/235089.
 #define MAYBE_CanSetupLegacyCall DISABLED_CanSetupLegacyCall
 #elif defined(OS_LINUX) && !defined(OS_CHROMEOS) && defined(ARCH_CPU_ARM_FAMILY)
 // Timing out on ARM linux, see http://crbug.com/240373
 #define MAYBE_CanSetupLegacyCall DISABLED_CanSetupLegacyCall
 #else
 #define MAYBE_CanSetupLegacyCall CanSetupLegacyCall
@@ skipping 239 matching lines @@
   GURL url(embedded_test_server()->GetURL("/media/peerconnection-call.html"));
   NavigateToURL(shell(), url);
 
   EXPECT_TRUE(ExecuteJavascript(
         base::StringPrintf("callAndEnsureAudioMutingWorks(%s);",
                            kForceIsac16K)));
   ExpectTitle("OK");
 }
 
 }  // namespace content