Add support for audio capture devices. Previously only output was supported.

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/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"
@@ skipping 48 matching lines @@
   DISALLOW_COPY_AND_ASSIGN(ReplaceContentClientRenderer);
 };
 
 namespace {
 
 class WebRTCMockResourceContext : public content::ResourceContext {
  public:
   WebRTCMockResourceContext() {}
   virtual ~WebRTCMockResourceContext() {}
   virtual void EnsureInitialized() const OVERRIDE {}
+
+ private:
+  DISALLOW_COPY_AND_ASSIGN(WebRTCMockResourceContext);
 };
 
 ACTION_P(QuitMessageLoop, loop_or_proxy) {
   loop_or_proxy->PostTask(FROM_HERE, new MessageLoop::QuitTask());
 }
 
 }  // end namespace
 
 WebRTCAudioDeviceTest::WebRTCAudioDeviceTest()
     : render_thread_(NULL), event_(false, false), audio_util_callback_(NULL) {
@@ skipping 43 matching lines @@
   // 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();
 }
 
 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_input_renderer_host_->OnChannelClosing();
   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;
 }
 
 void WebRTCAudioDeviceTest::OnGetHardwareInputSampleRate(double* sample_rate) {
   EXPECT_TRUE(audio_util_callback_);
@@ skipping 13 matching lines @@
     if (filter->OnMessageReceived(message))
       return true;
   }
 
   if (audio_render_host_.get()) {
     bool message_was_ok = false;
     if (audio_render_host_->OnMessageReceived(message, &message_was_ok))
       return true;
   }
 
+  if (audio_input_renderer_host_.get()) {
+    bool message_was_ok = false;
+    if (audio_input_renderer_host_->OnMessageReceived(message, &message_was_ok))
+      return true;
+  }
+
   bool handled = true;
   bool message_is_ok = true;
   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()
 
@@ skipping 27 matching lines @@
 #endif
 }
 
 WebRTCTransportImpl::WebRTCTransportImpl(webrtc::VoENetwork* network)
     : network_(network) {
 }
 
 WebRTCTransportImpl::~WebRTCTransportImpl() {}
 
 int WebRTCTransportImpl::SendPacket(int channel, const void* data, int len) {
-  ADD_FAILURE();  // We don't expect a call to this method in our tests.
   return network_->ReceivedRTPPacket(channel, data, len);
 }
 
 int WebRTCTransportImpl::SendRTCPPacket(int channel, const void* data,
                                         int len) {
   return network_->ReceivedRTCPPacket(channel, data, len);
 }