content/test/webrtc_audio_device_test.cc

   1 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
   2 // Use of this source code is governed by a BSD-style license that can be
   3 // found in the LICENSE file.
   4
   5 #include "content/test/webrtc_audio_device_test.h"
   6
   7 #include "base/bind.h"
   8 #include "base/bind_helpers.h"
   9 #include "base/compiler_specific.h"
  10 #include "base/file_util.h"
  11 #include "base/message_loop/message_loop.h"
  12 #include "base/run_loop.h"
  13 #include "base/synchronization/waitable_event.h"
  14 #include "base/test/test_timeouts.h"
  15 #include "content/browser/renderer_host/media/audio_input_renderer_host.h"
  16 #include "content/browser/renderer_host/media/audio_mirroring_manager.h"
  17 #include "content/browser/renderer_host/media/audio_renderer_host.h"
  18 #include "content/browser/renderer_host/media/media_stream_manager.h"
  19 #include "content/browser/renderer_host/media/mock_media_observer.h"
  20 #include "content/common/media/media_param_traits.h"
  21 #include "content/common/view_messages.h"
  22 #include "content/public/browser/browser_thread.h"
  23 #include "content/public/common/content_paths.h"
  24 #include "content/public/test/mock_resource_context.h"
  25 #include "content/public/test/test_browser_thread.h"
  26 #include "content/renderer/media/audio_input_message_filter.h"
  27 #include "content/renderer/media/audio_message_filter.h"
  28 #include "content/renderer/media/webrtc_audio_device_impl.h"
  29 #include "content/renderer/render_process.h"
  30 #include "content/renderer/render_thread_impl.h"
  31 #include "content/renderer/renderer_webkitplatformsupport_impl.h"
  32 #include "media/audio/audio_parameters.h"
  33 #include "media/base/audio_hardware_config.h"
  34 #include "net/url_request/url_request_test_util.h"
  35 #include "testing/gmock/include/gmock/gmock.h"
  36 #include "testing/gtest/include/gtest/gtest.h"
  37 #include "third_party/webrtc/voice_engine/include/voe_audio_processing.h"
  38 #include "third_party/webrtc/voice_engine/include/voe_base.h"
  39 #include "third_party/webrtc/voice_engine/include/voe_file.h"
  40 #include "third_party/webrtc/voice_engine/include/voe_network.h"
  41
  42 #if defined(OS_WIN)
  43 #include "base/win/scoped_com_initializer.h"
  44 #endif
  45
  46 using media::AudioParameters;
  47 using media::ChannelLayout;
  48 using testing::_;
  49 using testing::InvokeWithoutArgs;
  50 using testing::Return;
  51 using testing::StrEq;
  52
  53 namespace content {
  54
  55 // This class is a mock of the child process singleton which is needed
  56 // to be able to create a RenderThread object.
  57 class WebRTCMockRenderProcess : public RenderProcess {
  58  public:
  59   WebRTCMockRenderProcess() {}
  60   virtual ~WebRTCMockRenderProcess() {}
  61
  62   // RenderProcess implementation.
  63   virtual skia::PlatformCanvas* GetDrawingCanvas(
  64       TransportDIB** memory, const gfx::Rect& rect) OVERRIDE {
  65     return NULL;
  66   }
  67   virtual void ReleaseTransportDIB(TransportDIB* memory) OVERRIDE {}
  68   virtual bool UseInProcessPlugins() const OVERRIDE { return false; }
  69   virtual void AddBindings(int bindings) OVERRIDE {}
  70   virtual int GetEnabledBindings() const OVERRIDE { return 0; }
  71   virtual TransportDIB* CreateTransportDIB(size_t size) OVERRIDE {
  72     return NULL;
  73   }
  74   virtual void FreeTransportDIB(TransportDIB*) OVERRIDE {}
  75
  76  private:
  77   DISALLOW_COPY_AND_ASSIGN(WebRTCMockRenderProcess);
  78 };
  79
  80 // Utility scoped class to replace the global content client's renderer for the
  81 // duration of the test.
  82 class ReplaceContentClientRenderer {
  83  public:
  84   explicit ReplaceContentClientRenderer(ContentRendererClient* new_renderer) {
  85     saved_renderer_ = SetRendererClientForTesting(new_renderer);
  86   }
  87   ~ReplaceContentClientRenderer() {
  88     // Restore the original renderer.
  89     SetRendererClientForTesting(saved_renderer_);
  90   }
  91  private:
  92   ContentRendererClient* saved_renderer_;
  93   DISALLOW_COPY_AND_ASSIGN(ReplaceContentClientRenderer);
  94 };
  95
  96 class MockRTCResourceContext : public ResourceContext {
  97  public:
  98   MockRTCResourceContext() : test_request_context_(NULL) {}
  99   virtual ~MockRTCResourceContext() {}
 100
 101   void set_request_context(net::URLRequestContext* request_context) {
 102     test_request_context_ = request_context;
 103   }
 104
 105   // ResourceContext implementation:
 106   virtual net::HostResolver* GetHostResolver() OVERRIDE {
 107     return NULL;
 108   }
 109   virtual net::URLRequestContext* GetRequestContext() OVERRIDE {
 110     return test_request_context_;
 111   }
 112
 113  private:
 114   net::URLRequestContext* test_request_context_;
 115
 116   DISALLOW_COPY_AND_ASSIGN(MockRTCResourceContext);
 117 };
 118
 119 ACTION_P(QuitMessageLoop, loop_or_proxy) {
 120   loop_or_proxy->PostTask(FROM_HERE, base::MessageLoop::QuitClosure());
 121 }
 122
 123 WebRTCAudioDeviceTest::WebRTCAudioDeviceTest()
 124     : render_thread_(NULL), audio_hardware_config_(NULL),
 125       has_input_devices_(false), has_output_devices_(false) {
 126 }
 127
 128 WebRTCAudioDeviceTest::~WebRTCAudioDeviceTest() {}
 129
 130 void WebRTCAudioDeviceTest::SetUp() {
 131   // This part sets up a RenderThread environment to ensure that
 132   // RenderThread::current() (<=> TLS pointer) is valid.
 133   // Main parts are inspired by the RenderViewFakeResourcesTest.
 134   // Note that, the IPC part is not utilized in this test.
 135   saved_content_renderer_.reset(
 136       new ReplaceContentClientRenderer(&content_renderer_client_));
 137   mock_process_.reset(new WebRTCMockRenderProcess());
 138   ui_thread_.reset(
 139       new TestBrowserThread(BrowserThread::UI, base::MessageLoop::current()));
 140
 141   // Construct the resource context on the UI thread.
 142   resource_context_.reset(new MockRTCResourceContext);
 143
 144   static const char kThreadName[] = "RenderThread";
 145   ChildProcess::current()->io_message_loop()->PostTask(FROM_HERE,
 146       base::Bind(&WebRTCAudioDeviceTest::InitializeIOThread,
 147                  base::Unretained(this), kThreadName));
 148   WaitForIOThreadCompletion();
 149
 150   sandbox_was_enabled_ =
 151       RendererWebKitPlatformSupportImpl::SetSandboxEnabledForTesting(false);
 152   render_thread_ = new RenderThreadImpl(kThreadName);
 153 }
 154
 155 void WebRTCAudioDeviceTest::TearDown() {
 156   SetAudioHardwareConfig(NULL);
 157
 158   // Run any pending cleanup tasks that may have been posted to the main thread.
 159   base::RunLoop().RunUntilIdle();
 160
 161   // Kick of the cleanup process by closing the channel. This queues up
 162   // OnStreamClosed calls to be executed on the audio thread.
 163   ChildProcess::current()->io_message_loop()->PostTask(FROM_HERE,
 164       base::Bind(&WebRTCAudioDeviceTest::DestroyChannel,
 165                  base::Unretained(this)));
 166   WaitForIOThreadCompletion();
 167
 168   // When audio [input] render hosts are notified that the channel has
 169   // been closed, they post tasks to the audio thread to close the
 170   // AudioOutputController and once that's completed, a task is posted back to
 171   // the IO thread to actually delete the AudioEntry for the audio stream. Only
 172   // then is the reference to the audio manager released, so we wait for the
 173   // whole thing to be torn down before we finally uninitialize the io thread.
 174   WaitForAudioManagerCompletion();
 175
 176   ChildProcess::current()->io_message_loop()->PostTask(FROM_HERE,
 177       base::Bind(&WebRTCAudioDeviceTest::UninitializeIOThread,
 178                  base::Unretained((this))));
 179   WaitForIOThreadCompletion();
 180   mock_process_.reset();
 181   media_stream_manager_.reset();
 182   mirroring_manager_.reset();
 183   RendererWebKitPlatformSupportImpl::SetSandboxEnabledForTesting(
 184       sandbox_was_enabled_);
 185 }
 186
 187 bool WebRTCAudioDeviceTest::Send(IPC::Message* message) {
 188   return channel_->Send(message);
 189 }
 190
 191 void WebRTCAudioDeviceTest::SetAudioHardwareConfig(
 192     media::AudioHardwareConfig* hardware_config) {
 193   audio_hardware_config_ = hardware_config;
 194 }
 195
 196 void WebRTCAudioDeviceTest::InitializeIOThread(const char* thread_name) {
 197 #if defined(OS_WIN)
 198   // We initialize COM (STA) on our IO thread as is done in Chrome.
 199   // See BrowserProcessSubThread::Init.
 200   initialize_com_.reset(new base::win::ScopedCOMInitializer());
 201 #endif
 202
 203   // Set the current thread as the IO thread.
 204   io_thread_.reset(
 205       new TestBrowserThread(BrowserThread::IO, base::MessageLoop::current()));
 206
 207   // Populate our resource context.
 208   test_request_context_.reset(new net::TestURLRequestContext());
 209   MockRTCResourceContext* resource_context =
 210       static_cast<MockRTCResourceContext*>(resource_context_.get());
 211   resource_context->set_request_context(test_request_context_.get());
 212   media_internals_.reset(new MockMediaInternals());
 213
 214   // Create our own AudioManager, AudioMirroringManager and MediaStreamManager.
 215   audio_manager_.reset(media::AudioManager::Create());
 216   mirroring_manager_.reset(new AudioMirroringManager());
 217   media_stream_manager_.reset(new MediaStreamManager(audio_manager_.get()));
 218
 219   has_input_devices_ = audio_manager_->HasAudioInputDevices();
 220   has_output_devices_ = audio_manager_->HasAudioOutputDevices();
 221
 222   // Create an IPC channel that handles incoming messages on the IO thread.
 223   CreateChannel(thread_name);
 224 }
 225
 226 void WebRTCAudioDeviceTest::UninitializeIOThread() {
 227   resource_context_.reset();
 228
 229   test_request_context_.reset();
 230
 231 #if defined(OS_WIN)
 232   initialize_com_.reset();
 233 #endif
 234
 235   audio_manager_.reset();
 236 }
 237
 238 void WebRTCAudioDeviceTest::CreateChannel(const char* name) {
 239   DCHECK(BrowserThread::CurrentlyOn(BrowserThread::IO));
 240
 241   static const int kRenderProcessId = 1;
 242   audio_render_host_ = new AudioRendererHost(
 243       kRenderProcessId, audio_manager_.get(), mirroring_manager_.get(),
 244       media_internals_.get(), media_stream_manager_.get());
 245   audio_render_host_->OnChannelConnected(base::GetCurrentProcId());
 246
 247   audio_input_renderer_host_ = new AudioInputRendererHost(
 248       audio_manager_.get(), media_stream_manager_.get(),
 249       mirroring_manager_.get());
 250   audio_input_renderer_host_->OnChannelConnected(base::GetCurrentProcId());
 251
 252   channel_.reset(new IPC::Channel(name, IPC::Channel::MODE_SERVER, this));
 253   ASSERT_TRUE(channel_->Connect());
 254
 255   audio_render_host_->OnFilterAdded(channel_.get());
 256   audio_input_renderer_host_->OnFilterAdded(channel_.get());
 257 }
 258
 259 void WebRTCAudioDeviceTest::DestroyChannel() {
 260   DCHECK(BrowserThread::CurrentlyOn(BrowserThread::IO));
 261   audio_render_host_->OnChannelClosing();
 262   audio_render_host_->OnFilterRemoved();
 263   audio_input_renderer_host_->OnChannelClosing();
 264   audio_input_renderer_host_->OnFilterRemoved();
 265   channel_.reset();
 266   audio_render_host_ = NULL;
 267   audio_input_renderer_host_ = NULL;
 268 }
 269
 270 void WebRTCAudioDeviceTest::OnGetAudioHardwareConfig(
 271     AudioParameters* input_params, AudioParameters* output_params) {
 272   ASSERT_TRUE(audio_hardware_config_);
 273   *input_params = audio_hardware_config_->GetInputConfig();
 274   *output_params = audio_hardware_config_->GetOutputConfig();
 275 }
 276
 277 // IPC::Listener implementation.
 278 bool WebRTCAudioDeviceTest::OnMessageReceived(const IPC::Message& message) {
 279   if (render_thread_) {
 280     IPC::ChannelProxy::MessageFilter* filter =
 281         render_thread_->audio_input_message_filter();
 282     if (filter->OnMessageReceived(message))
 283       return true;
 284
 285     filter = render_thread_->audio_message_filter();
 286     if (filter->OnMessageReceived(message))
 287       return true;
 288   }
 289
 290   if (audio_render_host_.get()) {
 291     bool message_was_ok = false;
 292     if (audio_render_host_->OnMessageReceived(message, &message_was_ok))
 293       return true;
 294   }
 295
 296   if (audio_input_renderer_host_.get()) {
 297     bool message_was_ok = false;
 298     if (audio_input_renderer_host_->OnMessageReceived(message, &message_was_ok))
 299       return true;
 300   }
 301
 302   bool handled ALLOW_UNUSED = true;
 303   bool message_is_ok = true;
 304   IPC_BEGIN_MESSAGE_MAP_EX(WebRTCAudioDeviceTest, message, message_is_ok)
 305     IPC_MESSAGE_HANDLER(ViewHostMsg_GetAudioHardwareConfig,
 306                         OnGetAudioHardwareConfig)
 307     IPC_MESSAGE_UNHANDLED(handled = false)
 308   IPC_END_MESSAGE_MAP_EX()
 309
 310   EXPECT_TRUE(message_is_ok);
 311
 312   return true;
 313 }
 314
 315 // Posts a final task to the IO message loop and waits for completion.
 316 void WebRTCAudioDeviceTest::WaitForIOThreadCompletion() {
 317   WaitForMessageLoopCompletion(
 318       ChildProcess::current()->io_message_loop()->message_loop_proxy().get());
 319 }
 320
 321 void WebRTCAudioDeviceTest::WaitForAudioManagerCompletion() {
 322   if (audio_manager_)
 323     WaitForMessageLoopCompletion(audio_manager_->GetMessageLoop().get());
 324 }
 325
 326 void WebRTCAudioDeviceTest::WaitForMessageLoopCompletion(
 327     base::MessageLoopProxy* loop) {
 328   base::WaitableEvent* event = new base::WaitableEvent(false, false);
 329   loop->PostTask(FROM_HERE, base::Bind(&base::WaitableEvent::Signal,
 330                  base::Unretained(event)));
 331   if (event->TimedWait(TestTimeouts::action_max_timeout())) {
 332     delete event;
 333   } else {
 334     // Don't delete the event object in case the message ever gets processed.
 335     // If we do, we will crash the test process.
 336     ADD_FAILURE() << "Failed to wait for message loop";
 337   }
 338 }
 339
 340 std::string WebRTCAudioDeviceTest::GetTestDataPath(
 341     const base::FilePath::StringType& file_name) {
 342   base::FilePath path;
 343   EXPECT_TRUE(PathService::Get(DIR_TEST_DATA, &path));
 344   path = path.Append(file_name);
 345   EXPECT_TRUE(base::PathExists(path));
 346 #if defined(OS_WIN)
 347   return WideToUTF8(path.value());
 348 #else
 349   return path.value();
 350 #endif
 351 }
 352
 353 WebRTCTransportImpl::WebRTCTransportImpl(webrtc::VoENetwork* network)
 354     : network_(network) {
 355 }
 356
 357 WebRTCTransportImpl::~WebRTCTransportImpl() {}
 358
 359 int WebRTCTransportImpl::SendPacket(int channel, const void* data, int len) {
 360   return network_->ReceivedRTPPacket(channel, data, len);
 361 }
 362
 363 int WebRTCTransportImpl::SendRTCPPacket(int channel, const void* data,
 364                                         int len) {
 365   return network_->ReceivedRTCPPacket(channel, data, len);
 366 }
 367
 368 }  // namespace content