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_hardware.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 86 matching lines @@
       FROM_HERE,
       base::Bind(&SetupTask::InitializeIOThread, new SetupTask(this),
                  kThreadName));
   WaitForIOThreadCompletion();
 
   render_thread_ = new RenderThreadImpl(kThreadName);
   mock_process_->set_main_thread(render_thread_);
 }
 
 void WebRTCAudioDeviceTest::TearDown() {
+  SetAudioUtilCallback(NULL);
+
   ChildProcess::current()->io_message_loop()->PostTask(
       FROM_HERE,
       base::Bind(&SetupTask::UninitializeIOThread, new SetupTask(this)));
-  WaitForIOThreadCompletion();
+  EXPECT_TRUE(event_.TimedWait(
+      base::TimeDelta::FromMilliseconds(TestTimeouts::action_timeout_ms())));
   mock_process_.reset();
 }
 
 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.
+  audio_hardware::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());
   media_observer_.reset(new MockMediaObserver());
   resource_context_->set_media_observer(media_observer_.get());
   media_stream_manager_.reset(new media_stream::MediaStreamManager());
   resource_context_->set_media_stream_manager(media_stream_manager_.get());
 
   CreateChannel(thread_name, resource_context_.get());
 }
 
 void WebRTCAudioDeviceTest::UninitializeIOThread() {
   DestroyChannel();
   resource_context_.reset();
   media_stream_manager_.reset();
   test_request_context_ = NULL;
   initialize_com_.reset();
+  event_.Signal();
 }
 
 void WebRTCAudioDeviceTest::CreateChannel(
     const char* name,
     content::ResourceContext* resource_context) {
   DCHECK(content::BrowserThread::CurrentlyOn(content::BrowserThread::IO));
   audio_render_host_ = new AudioRendererHost(resource_context);
   audio_render_host_->OnChannelConnected(base::GetCurrentProcId());
 
   audio_input_renderer_host_ = new AudioInputRendererHost(resource_context);
   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));
   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(double* sample_rate) {
   EXPECT_TRUE(audio_util_callback_);
   *sample_rate = audio_util_callback_ ?
       audio_util_callback_->GetAudioHardwareSampleRate() : 0.0;
 }
@@ skipping 34 matching lines @@
   IPC_BEGIN_MESSAGE_MAP_EX(WebRTCAudioDeviceTest, message, message_is_ok)
     IPC_MESSAGE_HANDLER(ViewHostMsg_GetHardwareSampleRate,
                         OnGetHardwareSampleRate)
     IPC_MESSAGE_HANDLER(ViewHostMsg_GetHardwareInputSampleRate,
                         OnGetHardwareInputSampleRate)
     IPC_MESSAGE_UNHANDLED(handled = false)
   IPC_END_MESSAGE_MAP_EX()
 
   EXPECT_TRUE(message_is_ok);
 
-  // We leave a DLOG as a hint to the developer in case important IPC messages
-  // are being dropped.
-  DLOG_IF(WARNING, !handled) << "Unhandled IPC message";
-
   return true;
 }
 
 // Posts a final task to the IO message loop and waits for completion.
 void WebRTCAudioDeviceTest::WaitForIOThreadCompletion() {
   ChildProcess::current()->io_message_loop()->PostTask(
       FROM_HERE, new base::SignalingTask(&event_));
   EXPECT_TRUE(event_.TimedWait(
       base::TimeDelta::FromMilliseconds(TestTimeouts::action_timeout_ms())));
 }
@@ skipping 18 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);
 }