Refactor the Get*Hardware* routines a bit.

content/test/webrtc_audio_device_test.cc

 // Copyright (c) 2011 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/file_util.h"
 #include "base/message_loop.h"
 #include "base/synchronization/waitable_event.h"
 #include "base/test/signaling_task.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/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_device.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/test/test_browser_thread.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"
@@ skipping 91 matching lines @@
   render_thread_ = new RenderThreadImpl(kThreadName);
   mock_process_->set_main_thread(render_thread_);
 }
 
 void WebRTCAudioDeviceTest::TearDown() {
   ChildProcess::current()->io_message_loop()->PostTask(
       FROM_HERE,
       base::Bind(&SetupTask::UninitializeIOThread, new SetupTask(this)));
   WaitForIOThreadCompletion();
   mock_process_.reset();
+  SetAudioUtilCallback(NULL);
 }
 
 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.
+  AudioHardware::ResetCache();
+  audio_util_callback_ = callback;
+}
+
 void WebRTCAudioDeviceTest::InitializeIOThread(const char* thread_name) {
   // We initialize COM (STA) on our IO thread as is done in Chrome.
   // See BrowserProcessSubThread::Init.
   initialize_com_.reset(new ScopedCOMInitializer());
 
   // Set the current thread as the IO thread.
   io_thread_.reset(new content::TestBrowserThread(content::BrowserThread::IO,
                                                   MessageLoop::current()));
   test_request_context_ = new TestURLRequestContext();
   resource_context_->set_request_context(test_request_context_.get());
@@ skipping 123 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);
 }