Move a bunch of code in content\renderer to the content namespace.

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"
@@ skipping 27 matching lines @@
 
 #if defined(OS_WIN)
 #include "base/win/scoped_com_initializer.h"
 #endif
 
 using testing::_;
 using testing::InvokeWithoutArgs;
 using testing::Return;
 using testing::StrEq;
 
+namespace content {
+
 // This class is a mock of the child process singleton which is needed
 // to be able to create a RenderThread object.
 class WebRTCMockRenderProcess : public RenderProcess {
  public:
   WebRTCMockRenderProcess() {}
   virtual ~WebRTCMockRenderProcess() {}
 
   // RenderProcess implementation.
   virtual skia::PlatformCanvas* GetDrawingCanvas(
       TransportDIB** memory, const gfx::Rect& rect) OVERRIDE {
     return NULL;
   }
   virtual void ReleaseTransportDIB(TransportDIB* memory) OVERRIDE {}
   virtual bool UseInProcessPlugins() const OVERRIDE { return false; }
   virtual void AddBindings(int bindings) OVERRIDE {}
   virtual int GetEnabledBindings() const { return 0; }
   virtual TransportDIB* CreateTransportDIB(size_t size)  { return NULL; }
   virtual void FreeTransportDIB(TransportDIB*) {}
 
  private:
   DISALLOW_COPY_AND_ASSIGN(WebRTCMockRenderProcess);
 };
 
 // Utility scoped class to replace the global content client's renderer for the
 // duration of the test.
 class ReplaceContentClientRenderer {
  public:
-  explicit ReplaceContentClientRenderer(
-      content::ContentRendererClient* new_renderer) {
-    saved_renderer_ = content::GetContentClient()->renderer();
-    content::GetContentClient()->set_renderer_for_testing(new_renderer);
+  explicit ReplaceContentClientRenderer(ContentRendererClient* new_renderer) {
+    saved_renderer_ = GetContentClient()->renderer();
+    GetContentClient()->set_renderer_for_testing(new_renderer);
   }
   ~ReplaceContentClientRenderer() {
     // Restore the original renderer.
-    content::GetContentClient()->set_renderer_for_testing(saved_renderer_);
+    GetContentClient()->set_renderer_for_testing(saved_renderer_);
   }
  private:
-  content::ContentRendererClient* saved_renderer_;
+  ContentRendererClient* saved_renderer_;
   DISALLOW_COPY_AND_ASSIGN(ReplaceContentClientRenderer);
 };
 
-namespace {
-
-class MockResourceContext : public content::ResourceContext {
+class MockRTCResourceContext : public ResourceContext {
  public:
-  MockResourceContext() : test_request_context_(NULL) {}
-  virtual ~MockResourceContext() {}
+  MockRTCResourceContext() : test_request_context_(NULL) {}
+  virtual ~MockRTCResourceContext() {}
 
   void set_request_context(net::URLRequestContext* request_context) {
     test_request_context_ = request_context;
   }
 
   // ResourceContext implementation:
   virtual net::HostResolver* GetHostResolver() OVERRIDE {
     return NULL;
   }
   virtual net::URLRequestContext* GetRequestContext() OVERRIDE {
     return test_request_context_;
   }
 
  private:
   net::URLRequestContext* test_request_context_;
 
-  DISALLOW_COPY_AND_ASSIGN(MockResourceContext);
+  DISALLOW_COPY_AND_ASSIGN(MockRTCResourceContext);
 };
 
 ACTION_P(QuitMessageLoop, loop_or_proxy) {
   loop_or_proxy->PostTask(FROM_HERE, MessageLoop::QuitClosure());
 }
 
-}  // namespace
-
 WebRTCAudioDeviceTest::WebRTCAudioDeviceTest()
     : render_thread_(NULL), audio_util_callback_(NULL),
       has_input_devices_(false), has_output_devices_(false) {
 }
 
 WebRTCAudioDeviceTest::~WebRTCAudioDeviceTest() {}
 
 void WebRTCAudioDeviceTest::SetUp() {
   // This part sets up a RenderThread environment to ensure that
   // RenderThread::current() (<=> TLS pointer) is valid.
   // Main parts are inspired by the RenderViewFakeResourcesTest.
   // Note that, the IPC part is not utilized in this test.
   saved_content_renderer_.reset(
       new ReplaceContentClientRenderer(&content_renderer_client_));
   mock_process_.reset(new WebRTCMockRenderProcess());
-  ui_thread_.reset(new content::TestBrowserThread(content::BrowserThread::UI,
-                                                  MessageLoop::current()));
+  ui_thread_.reset(new TestBrowserThread(BrowserThread::UI,
+                                         MessageLoop::current()));
 
   // Create our own AudioManager and MediaStreamManager.
   audio_manager_.reset(media::AudioManager::Create());
   media_stream_manager_.reset(
       new media_stream::MediaStreamManager(audio_manager_.get()));
 
   // Construct the resource context on the UI thread.
-  resource_context_.reset(new MockResourceContext);
+  resource_context_.reset(new MockRTCResourceContext);
 
   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);
@@ skipping 43 matching lines @@
 }
 
 void WebRTCAudioDeviceTest::InitializeIOThread(const char* thread_name) {
 #if defined(OS_WIN)
   // We initialize COM (STA) on our IO thread as is done in Chrome.
   // See BrowserProcessSubThread::Init.
   initialize_com_.reset(new base::win::ScopedCOMInitializer());
 #endif
 
   // Set the current thread as the IO thread.
-  io_thread_.reset(new content::TestBrowserThread(content::BrowserThread::IO,
-                                                  MessageLoop::current()));
+  io_thread_.reset(new TestBrowserThread(BrowserThread::IO,
+                                         MessageLoop::current()));
 
   // Populate our resource context.
   test_request_context_.reset(new TestURLRequestContext());
-  MockResourceContext* resource_context =
-      static_cast<MockResourceContext*>(resource_context_.get());
+  MockRTCResourceContext* resource_context =
+      static_cast<MockRTCResourceContext*>(resource_context_.get());
   resource_context->set_request_context(test_request_context_.get());
   media_observer_.reset(new MockMediaObserver());
 
   has_input_devices_ = audio_manager_->HasAudioInputDevices();
   has_output_devices_ = audio_manager_->HasAudioOutputDevices();
 
   // Create an IPC channel that handles incoming messages on the IO thread.
   CreateChannel(thread_name);
 }
 
 void WebRTCAudioDeviceTest::UninitializeIOThread() {
   resource_context_.reset();
 
   test_request_context_.reset();
 
 #if defined(OS_WIN)
   initialize_com_.reset();
 #endif
 }
 
 void WebRTCAudioDeviceTest::CreateChannel(const char* name) {
-  DCHECK(content::BrowserThread::CurrentlyOn(content::BrowserThread::IO));
+  DCHECK(BrowserThread::CurrentlyOn(BrowserThread::IO));
   audio_render_host_ = new AudioRendererHost(
       audio_manager_.get(), media_observer_.get());
   audio_render_host_->OnChannelConnected(base::GetCurrentProcId());
 
   audio_input_renderer_host_ = new AudioInputRendererHost(
       audio_manager_.get(), media_stream_manager_.get());
   audio_input_renderer_host_->OnChannelConnected(base::GetCurrentProcId());
 
   channel_.reset(new IPC::Channel(name, IPC::Channel::MODE_SERVER, this));
   ASSERT_TRUE(channel_->Connect());
 
   audio_render_host_->OnFilterAdded(channel_.get());
   audio_input_renderer_host_->OnFilterAdded(channel_.get());
 }
 
 void WebRTCAudioDeviceTest::DestroyChannel() {
-  DCHECK(content::BrowserThread::CurrentlyOn(content::BrowserThread::IO));
+  DCHECK(BrowserThread::CurrentlyOn(BrowserThread::IO));
   audio_render_host_->OnChannelClosing();
   audio_render_host_->OnFilterRemoved();
   audio_input_renderer_host_->OnChannelClosing();
   audio_input_renderer_host_->OnFilterRemoved();
   channel_.reset();
   audio_render_host_ = NULL;
   audio_input_renderer_host_ = NULL;
 }
 
 void WebRTCAudioDeviceTest::OnGetHardwareSampleRate(int* sample_rate) {
@@ skipping 80 matching lines @@
   } else {
     // Don't delete the event object in case the message ever gets processed.
     // If we do, we will crash the test process.
     ADD_FAILURE() << "Failed to wait for message loop";
   }
 }
 
 std::string WebRTCAudioDeviceTest::GetTestDataPath(
     const FilePath::StringType& file_name) {
   FilePath path;
-  EXPECT_TRUE(PathService::Get(content::DIR_TEST_DATA, &path));
+  EXPECT_TRUE(PathService::Get(DIR_TEST_DATA, &path));
   path = path.Append(file_name);
   EXPECT_TRUE(file_util::PathExists(path));
 #ifdef OS_WIN
   return WideToUTF8(path.value());
 #else
   return path.value();
 #endif
 }
 
 WebRTCTransportImpl::WebRTCTransportImpl(webrtc::VoENetwork* network)
     : network_(network) {
 }
 
 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);
 }
+
+}  // namespace content