Make content::ResourceContext be a real interface like the rest of the Content API (i.e. don't ha...

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/mock_resource_context.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/browser/resource_context.h"
 #include "content/common/view_messages.h"
 #include "content/public/browser/browser_thread.h"
 #include "content/public/common/content_paths.h"
 #include "content/renderer/media/audio_hardware.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
@@ skipping 47 matching lines @@
     // Restore the original renderer.
     content::GetContentClient()->set_renderer(saved_renderer_);
   }
  private:
   content::ContentRendererClient* saved_renderer_;
   DISALLOW_COPY_AND_ASSIGN(ReplaceContentClientRenderer);
 };
 
 namespace {
 
-class WebRTCMockResourceContext : public content::ResourceContext {
- public:
-  WebRTCMockResourceContext() {}
-  virtual ~WebRTCMockResourceContext() {}
-  virtual void EnsureInitialized() const OVERRIDE {}
-
- private:
-  DISALLOW_COPY_AND_ASSIGN(WebRTCMockResourceContext);
-};
-
 ACTION_P(QuitMessageLoop, loop_or_proxy) {
   loop_or_proxy->PostTask(FROM_HERE, MessageLoop::QuitClosure());
 }
 
 }  // end namespace
 
 WebRTCAudioDeviceTest::WebRTCAudioDeviceTest()
     : render_thread_(NULL), audio_util_callback_(NULL) {
 }
 
 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(&mock_content_renderer_client_));
   mock_process_.reset(new WebRTCMockRenderProcess());
   ui_thread_.reset(new content::TestBrowserThread(content::BrowserThread::UI,
                                                   MessageLoop::current()));
 
   // Construct the resource context on the UI thread.
-  resource_context_.reset(new WebRTCMockResourceContext());
+  resource_context_.reset(new content::MockResourceContext(NULL));
 
   static const char kThreadName[] = "RenderThread";
   ChildProcess::current()->io_message_loop()->PostTask(FROM_HERE,
       base::Bind(&WebRTCAudioDeviceTest::InitializeIOThread,
                  base::Unretained(this), kThreadName));
   WaitForIOThreadCompletion();
 
   render_thread_ = new RenderThreadImpl(kThreadName);
 }
 
@@ skipping 201 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);
 }