Disabling WebRTC call tests on Android ASAN.

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 820dbd9b6973fdddb52c7d481a3593792b624eea..cd1d8b63c2dc74cbb98d6635e1938fd97055c28d 100644
--- a/content/browser/media/webrtc_browsertest.cc
+++ b/content/browser/media/webrtc_browsertest.cc
@@ -4,27 +4,18 @@
 
 #include "base/command_line.h"
 #include "base/file_util.h"
-#include "base/process/process_handle.h"
-#include "base/strings/string_number_conversions.h"
 #include "base/strings/stringprintf.h"
 #include "base/threading/platform_thread.h"
-#include "base/values.h"
-#include "content/browser/media/webrtc_internals.h"
 #include "content/browser/web_contents/web_contents_impl.h"
 #include "content/public/common/content_switches.h"
 #include "content/public/test/browser_test_utils.h"
 #include "content/public/test/content_browser_test_utils.h"
 #include "content/public/test/test_utils.h"
-#include "content/shell/browser/shell.h"
 #include "content/test/webrtc_content_browsertest_base.h"
 #include "media/audio/audio_manager.h"
 #include "media/base/media_switches.h"
 #include "net/test/embedded_test_server/embedded_test_server.h"
 
-#if defined(OS_WIN)
-#include "base/win/windows_version.h"
-#endif
-
 namespace {
 
 #if defined (OS_ANDROID) || defined(THREAD_SANITIZER)
@@ -35,29 +26,21 @@ const char kUseLenientAudioChecking[] = "true";
 const char kUseLenientAudioChecking[] = "false";
 #endif
 
-const int kExpectedConsumerId = 0;
-
-// Get the ID for the render process host when there should only be one.
-bool GetRenderProcessHostId(base::ProcessId* id) {
-  content::RenderProcessHost::iterator it(
-      content::RenderProcessHost::AllHostsIterator());
-  *id = base::GetProcId(it.GetCurrentValue()->GetHandle());
-  EXPECT_NE(base::kNullProcessId, *id);
-  if (*id == base::kNullProcessId)
-    return false;
-  it.Advance();
-  EXPECT_TRUE(it.IsAtEnd());
-  return it.IsAtEnd();
-}
-
 }  // namespace
 
 namespace content {
 
-class WebRtcBrowserTest : public WebRtcContentBrowserTest {
+#if defined(OS_ANDROID) && defined(ADDRESS_SANITIZER)
+// Renderer crashes under Android ASAN: https://crbug.com/408496.
+#define MAYBE_WebRtcBrowserTest DISABLED_WebRtcBrowserTest
+#else
+#define MAYBE_WebRtcBrowserTest WebRtcBrowserTest
+#endif
+
+class MAYBE_WebRtcBrowserTest : public WebRtcContentBrowserTest {
  public:
-  WebRtcBrowserTest() {}
-  virtual ~WebRtcBrowserTest() {}
+  MAYBE_WebRtcBrowserTest() {}
+  virtual ~MAYBE_WebRtcBrowserTest() {}
 
   // Convenience function since most peerconnection-call.html tests just load
   // the page, kick off some javascript and wait for the title to change to OK.
@@ -91,14 +74,6 @@ class WebRtcBrowserTest : public WebRtcContentBrowserTest {
 
     MakeTypicalPeerConnectionCall(javascript);
   }
-
-  void DisableOpusIfOnAndroid() {
-#if defined(OS_ANDROID)
-    // Always force iSAC 16K on Android for now (Opus is broken).
-    EXPECT_EQ("isac-forced",
-              ExecuteJavascriptAndReturnResult("forceIsac16KInSdp();"));
-#endif
-  }
 };
 
 #if defined(OS_LINUX) && !defined(OS_CHROMEOS) && defined(ARCH_CPU_ARM_FAMILY)
@@ -113,7 +88,8 @@ class WebRtcBrowserTest : public WebRtcContentBrowserTest {
 
 // These tests will make a complete PeerConnection-based call and verify that
 // video is playing for the call.
-IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest, MAYBE_CanSetupDefaultVideoCall) {
+IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest,
+                       MAYBE_CanSetupDefaultVideoCall) {
   MakeTypicalPeerConnectionCall(
       "callAndExpectResolution({video: true}, 640, 480);");
 }
@@ -126,7 +102,7 @@ IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest, MAYBE_CanSetupDefaultVideoCall) {
 #define MAYBE_CanSetupVideoCallWith1To1AspectRatio \
   CanSetupVideoCallWith1To1AspectRatio
 #endif
-IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest,
+IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest,
                        MAYBE_CanSetupVideoCallWith1To1AspectRatio) {
   const std::string javascript =
       "callAndExpectResolution({video: {mandatory: {minWidth: 320,"
@@ -146,7 +122,7 @@ IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest,
 #define MAYBE_CanSetupVideoCallWith16To9AspectRatio \
   CanSetupVideoCallWith16To9AspectRatio
 #endif
-IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest,
+IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest,
                        MAYBE_CanSetupVideoCallWith16To9AspectRatio) {
   const std::string javascript =
       "callAndExpectResolution({video: {mandatory: {minWidth: 640,"
@@ -162,7 +138,7 @@ IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest,
 #define MAYBE_CanSetupVideoCallWith4To3AspectRatio \
   CanSetupVideoCallWith4To3AspectRatio
 #endif
-IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest,
+IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest,
                        MAYBE_CanSetupVideoCallWith4To3AspectRatio) {
   const std::string javascript =
       "callAndExpectResolution({video: {mandatory: {minWidth: 960,"
@@ -178,7 +154,7 @@ IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest,
 #define MAYBE_CanSetupVideoCallAndDisableLocalVideo \
   CanSetupVideoCallAndDisableLocalVideo
 #endif
-IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest,
+IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest,
                        MAYBE_CanSetupVideoCallAndDisableLocalVideo) {
   const std::string javascript =
       "callAndDisableLocalVideo({video: true});";
@@ -192,18 +168,20 @@ IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest,
 #define MAYBE_CanSetupAudioAndVideoCall CanSetupAudioAndVideoCall
 #endif
 
-IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest, MAYBE_CanSetupAudioAndVideoCall) {
+IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest,
+                       MAYBE_CanSetupAudioAndVideoCall) {
   MakeTypicalPeerConnectionCall("call({video: true, audio: true});");
 }
 
-IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest, MANUAL_CanSetupCallAndSendDtmf) {
+IN_PROC_BROWSER_TEST_F(MAYBE_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.
 // http://crbug.com/293125.
-IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest,
+IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest,
                        DISABLED_CanMakeEmptyCallThenAddStreamsAndRenegotiate) {
   const char* kJavascript =
       "callEmptyThenAddOneStreamAndRenegotiate({video: true, audio: true});";
@@ -229,7 +207,7 @@ IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest,
 #define MAYBE_CanForwardRemoteStream720p CanForwardRemoteStream720p
 #endif
 #endif
-IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest, MAYBE_CanForwardRemoteStream) {
+IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest, MAYBE_CanForwardRemoteStream) {
 #if defined (OS_ANDROID)
   // This test fails on Nexus 5 devices.
   // TODO(henrika): see http://crbug.com/362437 and http://crbug.com/359389
@@ -241,7 +219,8 @@ IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest, MAYBE_CanForwardRemoteStream) {
       "callAndForwardRemoteStream({video: true, audio: false});");
 }
 
-IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest, MAYBE_CanForwardRemoteStream720p) {
+IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest,
+                       MAYBE_CanForwardRemoteStream720p) {
 #if defined (OS_ANDROID)
   // This test fails on Nexus 5 devices.
   // TODO(henrika): see http://crbug.com/362437 and http://crbug.com/359389
@@ -254,7 +233,7 @@ IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest, MAYBE_CanForwardRemoteStream720p) {
   MakeTypicalPeerConnectionCall(javascript);
 }
 
-IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest,
+IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest,
                        NoCrashWhenConnectChromiumSinkToRemoteTrack) {
   MakeTypicalPeerConnectionCall("ConnectChromiumSinkToRemoteAudioTrack();");
 }
@@ -271,26 +250,27 @@ IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest,
 #define MAYBE_CanSetupAudioAndVideoCallWithoutMsidAndBundle\
         CanSetupAudioAndVideoCallWithoutMsidAndBundle
 #endif
-IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest,
+IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest,
                        MAYBE_CanSetupAudioAndVideoCallWithoutMsidAndBundle) {
   MakeTypicalPeerConnectionCall("callWithoutMsidAndBundle();");
 }
 
 // This test will modify the SDP offer to an unsupported codec, which should
 // cause SetLocalDescription to fail.
-IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest, NegotiateUnsupportedVideoCodec) {
+IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest,
+                       NegotiateUnsupportedVideoCodec) {
   MakeTypicalPeerConnectionCall("negotiateUnsupportedVideoCodec();");
 }
 
 // This test will modify the SDP offer to use no encryption, which should
 // cause SetLocalDescription to fail.
-IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest, NegotiateNonCryptoCall) {
+IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest, NegotiateNonCryptoCall) {
   MakeTypicalPeerConnectionCall("negotiateNonCryptoCall();");
 }
 
 // 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) {
+IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest, NegotiateOfferWithBLine) {
   MakeTypicalPeerConnectionCall("negotiateOfferWithBLine();");
 }
 
@@ -303,14 +283,14 @@ IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest, NegotiateOfferWithBLine) {
 #define MAYBE_CanSetupLegacyCall CanSetupLegacyCall
 #endif
 
-IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest, MAYBE_CanSetupLegacyCall) {
+IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest, MAYBE_CanSetupLegacyCall) {
   MakeTypicalPeerConnectionCall("callWithLegacySdp();");
 }
 
 // This test will make a PeerConnection-based call and test an unreliable text
 // dataChannel.
 // TODO(mallinath) - Remove this test after rtp based data channel is disabled.
-IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest, CallWithDataOnly) {
+IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest, CallWithDataOnly) {
   MakeTypicalPeerConnectionCall("callWithDataOnly();");
 }
 
@@ -320,7 +300,7 @@ IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest, CallWithDataOnly) {
 #else
 #define MAYBE_CallWithSctpDataOnly CallWithSctpDataOnly
 #endif
-IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest, MAYBE_CallWithSctpDataOnly) {
+IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest, MAYBE_CallWithSctpDataOnly) {
   MakeTypicalPeerConnectionCall("callWithSctpDataOnly();");
 }
 
@@ -334,7 +314,7 @@ IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest, MAYBE_CallWithSctpDataOnly) {
 // This test will make a PeerConnection-based call and test an unreliable text
 // dataChannel and audio and video tracks.
 // TODO(mallinath) - Remove this test after rtp based data channel is disabled.
-IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest, DISABLED_CallWithDataAndMedia) {
+IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest, DISABLED_CallWithDataAndMedia) {
   MakeTypicalPeerConnectionCall("callWithDataAndMedia();");
 }
 
@@ -348,7 +328,7 @@ IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest, DISABLED_CallWithDataAndMedia) {
 #define MAYBE_CallWithSctpDataAndMedia CallWithSctpDataAndMedia
 #endif
 
-IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest,
+IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest,
                        MAYBE_CallWithSctpDataAndMedia) {
   MakeTypicalPeerConnectionCall("callWithSctpDataAndMedia();");
 }
@@ -363,7 +343,8 @@ IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest,
 
 // This test will make a PeerConnection-based call and test an unreliable text
 // dataChannel and later add an audio and video track.
-IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest, MAYBE_CallWithDataAndLaterAddMedia) {
+IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest,
+                       MAYBE_CallWithDataAndLaterAddMedia) {
   MakeTypicalPeerConnectionCall("callWithDataAndLaterAddMedia();");
 }
 
@@ -377,7 +358,8 @@ IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest, MAYBE_CallWithDataAndLaterAddMedia) {
 // This test will make a PeerConnection-based call and send a new Video
 // MediaStream that has been created based on a MediaStream created with
 // getUserMedia.
-IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest, MAYBE_CallWithNewVideoMediaStream) {
+IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest,
+                       MAYBE_CallWithNewVideoMediaStream) {
   MakeTypicalPeerConnectionCall("callWithNewVideoMediaStream();");
 }
 
@@ -387,51 +369,51 @@ IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest, MAYBE_CallWithNewVideoMediaStream) {
 // AudioTrack is added instead.
 // TODO(phoglund): This test is manual since not all buildbots has an audio
 // input.
-IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest, MANUAL_CallAndModifyStream) {
+IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest, MANUAL_CallAndModifyStream) {
   MakeTypicalPeerConnectionCall(
       "callWithNewVideoMediaStreamLaterSwitchToAudio();");
 }
 
-IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest, AddTwoMediaStreamsToOnePC) {
+IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest, AddTwoMediaStreamsToOnePC) {
   MakeTypicalPeerConnectionCall("addTwoMediaStreamsToOneConnection();");
 }
 
-IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest,
+IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest,
                        EstablishAudioVideoCallAndEnsureAudioIsPlaying) {
   MakeAudioDetectingPeerConnectionCall(base::StringPrintf(
       "callAndEnsureAudioIsPlaying(%s, {audio:true, video:true});",
       kUseLenientAudioChecking));
 }
 
-IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest,
+IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest,
                        EstablishAudioOnlyCallAndEnsureAudioIsPlaying) {
   MakeAudioDetectingPeerConnectionCall(base::StringPrintf(
       "callAndEnsureAudioIsPlaying(%s, {audio:true});",
       kUseLenientAudioChecking));
 }
 
-IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest,
+IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest,
                        EstablishAudioVideoCallAndVerifyRemoteMutingWorks) {
   MakeAudioDetectingPeerConnectionCall(base::StringPrintf(
       "callAndEnsureRemoteAudioTrackMutingWorks(%s);",
       kUseLenientAudioChecking));
 }
 
-IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest,
+IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest,
                        EstablishAudioVideoCallAndVerifyLocalMutingWorks) {
   MakeAudioDetectingPeerConnectionCall(base::StringPrintf(
       "callAndEnsureLocalAudioTrackMutingWorks(%s);",
       kUseLenientAudioChecking));
 }
 
-IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest,
+IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest,
                        EnsureLocalVideoMuteDoesntMuteAudio) {
   MakeAudioDetectingPeerConnectionCall(base::StringPrintf(
       "callAndEnsureLocalVideoMutingDoesntMuteAudio(%s);",
       kUseLenientAudioChecking));
 }
 
-IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest,
+IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest,
                        EnsureRemoteVideoMuteDoesntMuteAudio) {
   MakeAudioDetectingPeerConnectionCall(base::StringPrintf(
       "callAndEnsureRemoteVideoMutingDoesntMuteAudio(%s);",
@@ -446,177 +428,17 @@ IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest,
 #define MAYBE_EstablishAudioVideoCallAndVerifyUnmutingWorks\
         EstablishAudioVideoCallAndVerifyUnmutingWorks
 #endif
-IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest,
+IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest,
                        MAYBE_EstablishAudioVideoCallAndVerifyUnmutingWorks) {
   MakeAudioDetectingPeerConnectionCall(base::StringPrintf(
       "callAndEnsureAudioTrackUnmutingWorks(%s);", kUseLenientAudioChecking));
 }
 
-IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest, CallAndVerifyVideoMutingWorks) {
+IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest, CallAndVerifyVideoMutingWorks) {
   MakeTypicalPeerConnectionCall("callAndEnsureVideoTrackMutingWorks();");
 }
 
-#if defined(OS_WIN)
-#define IntToStringType base::IntToString16
-#else
-#define IntToStringType base::IntToString
-#endif
-
-#if defined(OS_LINUX) && !defined(OS_CHROMEOS) && defined(ARCH_CPU_ARM_FAMILY)
-// Timing out on ARM linux bot: http://crbug.com/238490
-#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
-// exists. The AEC dump is enabled through webrtc-internals. The HTML and
-// Javascript is bypassed since it would trigger a file picker dialog. Instead,
-// the dialog callback FileSelected() is invoked directly. In fact, there's
-// never a webrtc-internals page opened at all since that's not needed.
-IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest, MAYBE_CallWithAecDump) {
-  ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady());
-
-  // We must navigate somewhere first so that the render process is created.
-  NavigateToURL(shell(), GURL(""));
-
-  base::FilePath dump_file;
-  ASSERT_TRUE(CreateTemporaryFile(&dump_file));
-  base::DeleteFile(dump_file, false);
-
-  // This fakes the behavior of another open tab with webrtc-internals, and
-  // enabling AEC dump in that tab.
-  WebRTCInternals::GetInstance()->FileSelected(dump_file, -1, NULL);
-
-  GURL url(embedded_test_server()->GetURL("/media/peerconnection-call.html"));
-  NavigateToURL(shell(), url);
-  DisableOpusIfOnAndroid();
-  ExecuteJavascriptAndWaitForOk("call({video: true, audio: true});");
-
-  EXPECT_FALSE(base::PathExists(dump_file));
-
-  // Add file extensions that we expect to be added. The dump name will be
-  // <temporary path>.<render process id>.<consumer id>, for example
-  // "/tmp/.com.google.Chrome.Z6UC3P.12345.0".
-  base::ProcessId render_process_id = base::kNullProcessId;
-  EXPECT_TRUE(GetRenderProcessHostId(&render_process_id));
-  dump_file = dump_file.AddExtension(IntToStringType(render_process_id))
-                       .AddExtension(IntToStringType(kExpectedConsumerId));
-
-  EXPECT_TRUE(base::PathExists(dump_file));
-  int64 file_size = 0;
-  EXPECT_TRUE(base::GetFileSize(dump_file, &file_size));
-  EXPECT_GT(file_size, 0);
-
-  base::DeleteFile(dump_file, false);
-}
-
-// TODO(grunell): Add test for multiple dumps when re-use of
-// MediaStreamAudioProcessor in AudioCapturer has been removed.
-
-#if defined(OS_LINUX) && !defined(OS_CHROMEOS) && defined(ARCH_CPU_ARM_FAMILY)
-// Timing out on ARM linux bot: http://crbug.com/238490
-#define MAYBE_CallWithAecDumpEnabledThenDisabled DISABLED_CallWithAecDumpEnabledThenDisabled
-#else
-#define MAYBE_CallWithAecDumpEnabledThenDisabled CallWithAecDumpEnabledThenDisabled
-#endif
-
-// As above, but enable and disable dump before starting a call. The file should
-// be created, but should be empty.
-IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest,
-                       MAYBE_CallWithAecDumpEnabledThenDisabled) {
-  ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady());
-
-  // We must navigate somewhere first so that the render process is created.
-  NavigateToURL(shell(), GURL(""));
-
-  base::FilePath dump_file;
-  ASSERT_TRUE(CreateTemporaryFile(&dump_file));
-  base::DeleteFile(dump_file, false);
-
-  // This fakes the behavior of another open tab with webrtc-internals, and
-  // enabling AEC dump in that tab, then disabling it.
-  WebRTCInternals::GetInstance()->FileSelected(dump_file, -1, NULL);
-  WebRTCInternals::GetInstance()->DisableAecDump();
-
-  GURL url(embedded_test_server()->GetURL("/media/peerconnection-call.html"));
-  NavigateToURL(shell(), url);
-  DisableOpusIfOnAndroid();
-  ExecuteJavascriptAndWaitForOk("call({video: true, audio: true});");
-
-  // Add file extensions that we expect to be added.
-  base::ProcessId render_process_id = base::kNullProcessId;
-  EXPECT_TRUE(GetRenderProcessHostId(&render_process_id));
-  dump_file = dump_file.AddExtension(IntToStringType(render_process_id))
-                       .AddExtension(IntToStringType(kExpectedConsumerId));
-
-  EXPECT_FALSE(base::PathExists(dump_file));
-
-  base::DeleteFile(dump_file, false);
-}
-
-#if defined(OS_LINUX) && !defined(OS_CHROMEOS) && defined(ARCH_CPU_ARM_FAMILY)
-// Timing out on ARM linux bot: http://crbug.com/238490
-#define MAYBE_TwoCallsWithAecDump DISABLED_TwoCallsWithAecDump
-#else
-#define MAYBE_TwoCallsWithAecDump TwoCallsWithAecDump
-#endif
-
-IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest, MAYBE_TwoCallsWithAecDump) {
-  ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady());
-
-  // We must navigate somewhere first so that the render process is created.
-  NavigateToURL(shell(), GURL(""));
-
-  // Create a second window.
-  Shell* shell2 = CreateBrowser();
-  NavigateToURL(shell2, GURL(""));
-
-  base::FilePath dump_file;
-  ASSERT_TRUE(CreateTemporaryFile(&dump_file));
-  base::DeleteFile(dump_file, false);
-
-  // This fakes the behavior of another open tab with webrtc-internals, and
-  // enabling AEC dump in that tab.
-  WebRTCInternals::GetInstance()->FileSelected(dump_file, -1, NULL);
-
-  GURL url(embedded_test_server()->GetURL("/media/peerconnection-call.html"));
-
-  NavigateToURL(shell(), url);
-  NavigateToURL(shell2, url);
-  ExecuteJavascriptAndWaitForOk("call({video: true, audio: true});");
-  std::string result;
-  EXPECT_TRUE(ExecuteScriptAndExtractString(
-      shell2->web_contents(),
-      "call({video: true, audio: true});",
-      &result));
-  ASSERT_STREQ("OK", result.c_str());
-
-  EXPECT_FALSE(base::PathExists(dump_file));
-
-  RenderProcessHost::iterator it =
-      content::RenderProcessHost::AllHostsIterator();
-  for (; !it.IsAtEnd(); it.Advance()) {
-    base::ProcessId render_process_id =
-        base::GetProcId(it.GetCurrentValue()->GetHandle());
-    EXPECT_NE(base::kNullProcessId, render_process_id);
-
-    // Add file extensions that we expect to be added.
-    base::FilePath unique_dump_file =
-        dump_file.AddExtension(IntToStringType(render_process_id))
-                 .AddExtension(IntToStringType(kExpectedConsumerId));
-
-    EXPECT_TRUE(base::PathExists(unique_dump_file));
-    int64 file_size = 0;
-    EXPECT_TRUE(base::GetFileSize(unique_dump_file, &file_size));
-    EXPECT_GT(file_size, 0);
-
-    base::DeleteFile(unique_dump_file, false);
-  }
-}
-
-IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest, CreateOfferWithOfferOptions) {
+IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcBrowserTest, CreateOfferWithOfferOptions) {
   MakeTypicalPeerConnectionCall("testCreateOfferOptions();");
 }