Tab Audio Capture: Browser-side connect/disconnect functionality.

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 "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/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_hardware.h"
 #include "content/renderer/media/audio_input_message_filter.h"
@@ skipping 140 matching lines @@
   // then is the reference to the audio manager released, so we wait for the
   // whole thing to be torn down before we finally uninitialize the io thread.
   WaitForAudioManagerCompletion();
 
   ChildProcess::current()->io_message_loop()->PostTask(FROM_HERE,
       base::Bind(&WebRTCAudioDeviceTest::UninitializeIOThread,
                  base::Unretained((this))));
   WaitForIOThreadCompletion();
   mock_process_.reset();
   media_stream_manager_.reset();
+  mirroring_manager_.reset();
   audio_manager_.reset();
   RendererWebKitPlatformSupportImpl::SetSandboxEnabledForTesting(
       sandbox_was_enabled_);
 }
 
 bool WebRTCAudioDeviceTest::Send(IPC::Message* message) {
   return channel_->Send(message);
 }
 
 void WebRTCAudioDeviceTest::SetAudioUtilCallback(AudioUtilInterface* callback) {
@@ skipping 14 matching lines @@
   io_thread_.reset(new TestBrowserThread(BrowserThread::IO,
                                          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_observer_.reset(new MockMediaObserver());
 
-  // Create our own AudioManager and MediaStreamManager.
+  // 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.
   CreateChannel(thread_name);
 }
 
 void WebRTCAudioDeviceTest::UninitializeIOThread() {
   resource_context_.reset();
 
   test_request_context_.reset();
 
 #if defined(OS_WIN)
   initialize_com_.reset();
 #endif
 }
 
 void WebRTCAudioDeviceTest::CreateChannel(const char* name) {
   DCHECK(BrowserThread::CurrentlyOn(BrowserThread::IO));
+
+  static const int kRenderProcessId = 1;
   audio_render_host_ = new AudioRendererHost(
-      audio_manager_.get(), media_observer_.get());
+      kRenderProcessId, audio_manager_.get(), mirroring_manager_.get(),
+      media_observer_.get());
   audio_render_host_->OnChannelConnected(base::GetCurrentProcId());
 
   audio_input_renderer_host_ = new AudioInputRendererHost(
       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());
@@ skipping 121 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