Remove MockMediaInternals and incestuous usage.

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/message_loop.h"
 #include "base/run_loop.h"
 #include "base/synchronization/waitable_event.h"
 #include "base/test/test_timeouts.h"
+#include "content/browser/media/media_internals.h"
 #include "content/browser/renderer_host/media/audio_input_renderer_host.h"
 #include "content/browser/renderer_host/media/audio_mirroring_manager.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/media/media_param_traits.h"
 #include "content/common/view_messages.h"
 #include "content/public/browser/browser_thread.h"
 #include "content/public/browser/resource_context.h"
 #include "content/public/common/content_paths.h"
@@ skipping 252 matching lines @@
 
   // Set the current thread as the IO thread.
   io_thread_.reset(
       new TestBrowserThread(BrowserThread::IO, base::MessageLoop::current()));
 
   // Populate our resource context.
   test_request_context_.reset(new net::TestURLRequestContext());
   MockRTCResourceContext* resource_context =
       static_cast<MockRTCResourceContext*>(resource_context_.get());
   resource_context->set_request_context(test_request_context_.get());
-  media_internals_.reset(new MockMediaInternals());
 
   // Create our own AudioManager, AudioMirroringManager and MediaStreamManager.
   audio_manager_.reset(media::AudioManager::Create());
   mirroring_manager_.reset(new AudioMirroringManager());
   media_stream_manager_.reset(new MediaStreamManager(audio_manager_.get()));
 
   has_input_devices_ = audio_manager_->HasAudioInputDevices();
   has_output_devices_ = audio_manager_->HasAudioOutputDevices();
 
   // Create an IPC channel that handles incoming messages on the IO thread.
@@ skipping 12 matching lines @@
   audio_manager_.reset();
 }
 
 void MAYBE_WebRTCAudioDeviceTest::CreateChannel(const char* name) {
   DCHECK(BrowserThread::CurrentlyOn(BrowserThread::IO));
 
   channel_.reset(new IPC::Channel(name, IPC::Channel::MODE_SERVER, this));
   ASSERT_TRUE(channel_->Connect());
 
   static const int kRenderProcessId = 1;
-  audio_render_host_ = new TestAudioRendererHost(
-      kRenderProcessId, audio_manager_.get(), mirroring_manager_.get(),
-      media_internals_.get(), media_stream_manager_.get(), channel_.get());
+  audio_render_host_ = new TestAudioRendererHost(kRenderProcessId,
+                                                 audio_manager_.get(),
+                                                 mirroring_manager_.get(),
+                                                 MediaInternals::GetInstance(),
+                                                 media_stream_manager_.get(),
+                                                 channel_.get());
   audio_render_host_->set_peer_pid_for_testing(base::GetCurrentProcId());
 
   audio_input_renderer_host_ =
       new TestAudioInputRendererHost(audio_manager_.get(),
                                      media_stream_manager_.get(),
                                      mirroring_manager_.get(),
                                      NULL,
                                      channel_.get());
   audio_input_renderer_host_->set_peer_pid_for_testing(
       base::GetCurrentProcId());
@@ skipping 105 matching lines @@
 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