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"
@@ skipping 167 matching lines @@
 
   DISALLOW_COPY_AND_ASSIGN(MockRTCResourceContext);
 };
 
 ACTION_P(QuitMessageLoop, loop_or_proxy) {
   loop_or_proxy->PostTask(FROM_HERE, base::MessageLoop::QuitClosure());
 }
 
 MAYBE_WebRTCAudioDeviceTest::MAYBE_WebRTCAudioDeviceTest()
     : render_thread_(NULL), audio_hardware_config_(NULL),
       has_input_devices_(false), has_output_devices_(false) {

[tommi (sloooow) - chröme, 2013/11/22 11:55:31]

initialize media_internals_

[DaleCurtis, 2013/11/22 20:33:33]

No longer necessary.

 }
 
 MAYBE_WebRTCAudioDeviceTest::~MAYBE_WebRTCAudioDeviceTest() {}
 
 void MAYBE_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(
@@ skipping 78 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());
+  media_internals_ = MediaInternals::GetInstance();
 
   // 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 14 matching lines @@
 
 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());
+      media_internals_, media_stream_manager_.get(), channel_.get());

[acolwell GONE FROM CHROMIUM, 2013/11/22 19:12:48]

This appears to be the only use of media_internals_. Can't you just use
MediaInternals::GetInstance() here and remove the need for the member variable?

[DaleCurtis, 2013/11/22 20:33:33]

Done.

   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