Fix client-side phishing detection test flakiness and ChromeOS failure.

content/test/webrtc_audio_device_test.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 "content/test/webrtc_audio_device_test.h"
 
 #include "base/bind.h"
 #include "base/bind_helpers.h"
 #include "base/compiler_specific.h"
 #include "base/file_util.h"
 #include "base/message_loop.h"
 #include "base/synchronization/waitable_event.h"
 #include "base/test/test_timeouts.h"
 #include "base/win/scoped_com_initializer.h"
 #include "content/browser/renderer_host/media/audio_input_renderer_host.h"
 #include "content/browser/renderer_host/media/audio_renderer_host.h"
 #include "content/browser/renderer_host/media/media_stream_manager.h"
 #include "content/browser/renderer_host/media/mock_media_observer.h"
 #include "content/common/view_messages.h"
 #include "content/public/browser/browser_thread.h"
 #include "content/public/common/content_paths.h"
 #include "content/public/test/mock_resource_context.h"
+#include "content/public/test/test_browser_thread.h"
 #include "content/renderer/media/audio_hardware.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "content/renderer/render_process.h"
 #include "content/renderer/render_thread_impl.h"
-#include "content/public/test/test_browser_thread.h"
+#include "content/renderer/renderer_webkitplatformsupport_impl.h"
 #include "net/url_request/url_request_test_util.h"
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "third_party/webrtc/voice_engine/main/interface/voe_audio_processing.h"
 #include "third_party/webrtc/voice_engine/main/interface/voe_base.h"
 #include "third_party/webrtc/voice_engine/main/interface/voe_file.h"
 #include "third_party/webrtc/voice_engine/main/interface/voe_network.h"
 
 using base::win::ScopedCOMInitializer;
 using testing::_;
@@ skipping 92 matching lines @@
 
   // Construct the resource context on the UI thread.
   resource_context_.reset(new MockResourceContext);
 
   static const char kThreadName[] = "RenderThread";
   ChildProcess::current()->io_message_loop()->PostTask(FROM_HERE,
       base::Bind(&WebRTCAudioDeviceTest::InitializeIOThread,
                  base::Unretained(this), kThreadName));
   WaitForIOThreadCompletion();
 
+  sandbox_was_enabled_ =
+      RendererWebKitPlatformSupportImpl::SetSandboxEnabledForTesting(false);
   render_thread_ = new RenderThreadImpl(kThreadName);
 }
 
 void WebRTCAudioDeviceTest::TearDown() {
   SetAudioUtilCallback(NULL);
 
   // Run any pending cleanup tasks that may have been posted to the main thread.
   ChildProcess::current()->main_thread()->message_loop()->RunAllPending();
 
   // Kick of the cleanup process by closing the channel. This queues up
   // OnStreamClosed calls to be executed on the audio thread.
   ChildProcess::current()->io_message_loop()->PostTask(FROM_HERE,
       base::Bind(&WebRTCAudioDeviceTest::DestroyChannel,
                  base::Unretained(this)));
   WaitForIOThreadCompletion();
 
   // When audio [input] render hosts are notified that the channel has
   // been closed, they post tasks to the audio thread to close the
   // AudioOutputController and once that's completed, a task is posted back to
   // the IO thread to actually delete the AudioEntry for the audio stream. Only
   // then is the reference to the audio manager released, so we wait for the
   // whole thing to be torn down before we finally uninitialize the io thread.
   WaitForAudioManagerCompletion();
 
   ChildProcess::current()->io_message_loop()->PostTask(FROM_HERE,
       base::Bind(&WebRTCAudioDeviceTest::UninitializeIOThread,
                  base::Unretained((this))));
   WaitForIOThreadCompletion();
   mock_process_.reset();
+  RendererWebKitPlatformSupportImpl::SetSandboxEnabledForTesting(
+      sandbox_was_enabled_);
 }
 
 bool WebRTCAudioDeviceTest::Send(IPC::Message* message) {
   return channel_->Send(message);
 }
 
 void WebRTCAudioDeviceTest::SetAudioUtilCallback(AudioUtilInterface* callback) {
   // Invalidate any potentially cached values since the new callback should
   // be used for those queries.
   audio_hardware::ResetCache();
@@ skipping 170 matching lines @@
 WebRTCTransportImpl::~WebRTCTransportImpl() {}
 
 int WebRTCTransportImpl::SendPacket(int channel, const void* data, int len) {
   return network_->ReceivedRTPPacket(channel, data, len);
 }
 
 int WebRTCTransportImpl::SendRTCPPacket(int channel, const void* data,
                                         int len) {
   return network_->ReceivedRTCPPacket(channel, data, len);
 }