Move the device enumerate/open/close work to device thread from IO thread

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/renderer_host/media/audio_input_device_manager.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/renderer_host/media/video_capture_manager.h"
 #include "content/common/view_messages.h"
 #include "content/public/browser/browser_thread.h"
 #include "content/public/common/content_paths.h"
 #include "content/public/test/mock_resource_context.h"
 #include "content/public/test/test_browser_thread.h"
 #include "content/renderer/media/audio_device_factory.h"
 #include "content/renderer/media/audio_hardware.h"
 #include "content/renderer/media/audio_message_filter.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "content/renderer/render_process.h"
@@ skipping 14 matching lines @@
 using testing::StrEq;
 
 // This class is a mock of the child process singleton which is needed
 // to be able to create a RenderThread object.
 class WebRTCMockRenderProcess : public RenderProcess {
  public:
   WebRTCMockRenderProcess() {}
   virtual ~WebRTCMockRenderProcess() {}
 
   // RenderProcess implementation.
-  virtual skia::PlatformCanvas* GetDrawingCanvas(TransportDIB** memory,
-                                                 const gfx::Rect& rect) {
+  virtual skia::PlatformCanvas* GetDrawingCanvas(
+      TransportDIB** memory, const gfx::Rect& rect) OVERRIDE {
     return NULL;
   }
-  virtual void ReleaseTransportDIB(TransportDIB* memory) {}
-  virtual bool UseInProcessPlugins() const { return false; }
-  virtual void AddBindings(int bindings) {}
+  virtual void ReleaseTransportDIB(TransportDIB* memory) OVERRIDE {}
+  virtual bool UseInProcessPlugins() const OVERRIDE { return false; }
+  virtual void AddBindings(int bindings) OVERRIDE {}
   virtual int GetEnabledBindings() const { return 0; }
   virtual bool HasInitializedMediaLibrary() const { return false; }
 
  private:
   DISALLOW_COPY_AND_ASSIGN(WebRTCMockRenderProcess);
 };
 
 // Utility scoped class to replace the global content client's renderer for the
 // duration of the test.
 class ReplaceContentClientRenderer {
@@ skipping 120 matching lines @@
   // 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()));
 
   audio_manager_.reset(media::AudioManager::Create());
 
+  scoped_refptr<media_stream::AudioInputDeviceManager>
+      audio_input_device_manager(new media_stream::AudioInputDeviceManager(
+          audio_manager_.get()));
+  scoped_refptr<media_stream::VideoCaptureManager> video_capture_manager(
+      new media_stream::VideoCaptureManager());
+  media_stream_manager_.reset(new media_stream::MediaStreamManager(
+      audio_input_device_manager, video_capture_manager));
+
   // Populate our resource context.
   test_request_context_.reset(new TestURLRequestContext());
   MockResourceContext* resource_context =
       static_cast<MockResourceContext*>(resource_context_.get());
   resource_context->set_request_context(test_request_context_.get());
   media_observer_.reset(new MockMediaObserver());
 
   has_input_devices_ = audio_manager_->HasAudioInputDevices();
   has_output_devices_ = audio_manager_->HasAudioOutputDevices();
 
   // Create an IPC channel that handles incoming messages on the IO thread.
   CreateChannel(thread_name);
 }
 
 void WebRTCAudioDeviceTest::UninitializeIOThread() {
   resource_context_.reset();
 
+  media_stream_manager_.reset();
   audio_manager_.reset();
   test_request_context_.reset();
   initialize_com_.reset();
 }
 
 void WebRTCAudioDeviceTest::CreateChannel(const char* name) {
   DCHECK(content::BrowserThread::CurrentlyOn(content::BrowserThread::IO));
   audio_render_host_ = new AudioRendererHost(
       audio_manager_.get(), media_observer_.get());
   audio_render_host_->OnChannelConnected(base::GetCurrentProcId());
 
   audio_input_renderer_host_ = new AudioInputRendererHost(
-      resource_context_.get(), audio_manager_.get());
+      audio_manager_.get(), media_stream_manager_.get());
   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() {
@@ skipping 116 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);
 }