Ported media stream track enable-disable test to content module.

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 431 matching lines @@
 #endif
 
 IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, MAYBE_CanSetupAudioAndVideoCall) {
   MakeTypicalPeerConnectionCall("call({video: true, audio: true});");
 }
 
 IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, MANUAL_CanSetupCallAndSendDtmf) {
   MakeTypicalPeerConnectionCall("callAndSendDtmf('123,abc');");
 }
 
+// TODO(phoglund): this test fails because the peer connection state will be
+// stable in the second negotiation round rather than have-local-offer.

[perkj_chrome, 2014/01/08 08:55:51]

Somewhere it is a bug. 
See http://dev.w3.org/2011/webrtc/editor/webrtc.html and look at the state
transition diagram.

It seems like this has worked and has been broken
https://code.google.com/p/chromium/issues/detail?id=29312.

 IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest,
                        DISABLED_CanMakeEmptyCallThenAddStreamsAndRenegotiate) {
   const char* kJavascript =
       "callEmptyThenAddOneStreamAndRenegotiate({video: true, audio: true});";
   MakeTypicalPeerConnectionCall(kJavascript);
 }
 
 // Below 2 test will make a complete PeerConnection-based call between pc1 and
 // pc2, and then use the remote stream to setup a call between pc3 and pc4, and
 // then verify that video is received on pc3 and pc4.
@@ skipping 232 matching lines @@
   }
 
   ASSERT_TRUE(CommandLine::ForCurrentProcess()->HasSwitch(
       switches::kUseFakeDeviceForMediaStream))
           << "Must run with fake devices since the test will explicitly look "
           << "for the fake device signal.";
 
   MakeTypicalPeerConnectionCall("callAndEnsureAudioMutingWorks();");
 }
 
+IN_PROC_BROWSER_TEST_F(WebrtcBrowserTest, CallAndVerifyVideoMutingWorks) {
+  ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady());
+
+  GURL url(embedded_test_server()->GetURL("/media/peerconnection-call.html"));
+  NavigateToURL(shell(), url);
+
+  EXPECT_TRUE(ExecuteJavascript("callAndEnsureVideoMutingWorks();"));
+  ExpectTitle("OK");
+}
+
 #if defined(OS_WIN) || (defined(OS_LINUX) && !defined(OS_CHROMEOS) && defined(ARCH_CPU_ARM_FAMILY))
 // Timing out on ARM linux bot: http://crbug.com/238490
 // Failing on Windows: http://crbug.com/331035
 #define MAYBE_CallWithAecDump DISABLED_CallWithAecDump
 #else
 #define MAYBE_CallWithAecDump CallWithAecDump
 #endif
 
 // This tests will make a complete PeerConnection-based call, verify that
 // video is playing for the call, and verify that a non-empty AEC dump file
@@ skipping 62 matching lines @@
   EXPECT_TRUE(base::PathExists(dump_file));
   int64 file_size = 0;
   EXPECT_TRUE(base::GetFileSize(dump_file, &file_size));
   EXPECT_EQ(0, file_size);
 
   base::DeleteFile(dump_file, false);
 }
 
 
 }  // namespace content