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/browser/resource_context.h"
  24 #include "content/public/common/content_paths.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 class TestAudioRendererHost : public AudioRendererHost {
  81  public:
  82   TestAudioRendererHost(
  83       int render_process_id,
  84       media::AudioManager* audio_manager,
  85       AudioMirroringManager* mirroring_manager,
  86       MediaInternals* media_internals,
  87       MediaStreamManager* media_stream_manager,
  88       IPC::Channel* channel)
  89       : AudioRendererHost(render_process_id, audio_manager, mirroring_manager,
  90                           media_internals, media_stream_manager),
  91         channel_(channel) {}
  92   virtual bool Send(IPC::Message* message) OVERRIDE {
  93     if (channel_)
  94       return channel_->Send(message);
  95     return false;
  96   }
  97   void ResetChannel() {
  98     channel_ = NULL;
  99   }
 100
 101  protected:
 102   virtual ~TestAudioRendererHost() {}
 103
 104  private:
 105   IPC::Channel* channel_;
 106 };
 107
 108 class TestAudioInputRendererHost : public AudioInputRendererHost {
 109  public:
 110   TestAudioInputRendererHost(
 111       media::AudioManager* audio_manager,
 112       MediaStreamManager* media_stream_manager,
 113       AudioMirroringManager* audio_mirroring_manager,
 114       media::UserInputMonitor* user_input_monitor,
 115       IPC::Channel* channel)
 116       : AudioInputRendererHost(audio_manager, media_stream_manager,
 117                                audio_mirroring_manager, user_input_monitor),
 118         channel_(channel) {}
 119   virtual bool Send(IPC::Message* message) OVERRIDE {
 120     if (channel_)
 121       return channel_->Send(message);
 122     return false;
 123   }
 124   void ResetChannel() {
 125     channel_ = NULL;
 126   }
 127
 128  protected:
 129   virtual ~TestAudioInputRendererHost() {}
 130
 131  private:
 132   IPC::Channel* channel_;
 133 };
 134
 135 // Utility scoped class to replace the global content client's renderer for the
 136 // duration of the test.
 137 class ReplaceContentClientRenderer {
 138  public:
 139   explicit ReplaceContentClientRenderer(ContentRendererClient* new_renderer) {
 140     saved_renderer_ = SetRendererClientForTesting(new_renderer);
 141   }
 142   ~ReplaceContentClientRenderer() {
 143     // Restore the original renderer.
 144     SetRendererClientForTesting(saved_renderer_);
 145   }
 146  private:
 147   ContentRendererClient* saved_renderer_;
 148   DISALLOW_COPY_AND_ASSIGN(ReplaceContentClientRenderer);
 149 };
 150
 151 class MockRTCResourceContext : public ResourceContext {
 152  public:
 153   MockRTCResourceContext() : test_request_context_(NULL) {}
 154   virtual ~MockRTCResourceContext() {}
 155
 156   void set_request_context(net::URLRequestContext* request_context) {
 157     test_request_context_ = request_context;
 158   }
 159
 160   // ResourceContext implementation:
 161   virtual net::HostResolver* GetHostResolver() OVERRIDE {
 162     return NULL;
 163   }
 164   virtual net::URLRequestContext* GetRequestContext() OVERRIDE {
 165     return test_request_context_;
 166   }
 167
 168   virtual bool AllowMicAccess(const GURL& origin) OVERRIDE {
 169     return false;
 170   }
 171
 172   virtual bool AllowCameraAccess(const GURL& origin) OVERRIDE {
 173     return false;
 174   }
 175
 176  private:
 177   net::URLRequestContext* test_request_context_;
 178
 179   DISALLOW_COPY_AND_ASSIGN(MockRTCResourceContext);
 180 };
 181
 182 ACTION_P(QuitMessageLoop, loop_or_proxy) {
 183   loop_or_proxy->PostTask(FROM_HERE, base::MessageLoop::QuitClosure());
 184 }
 185
 186 MAYBE_WebRTCAudioDeviceTest::MAYBE_WebRTCAudioDeviceTest()
 187     : render_thread_(NULL), audio_hardware_config_(NULL),
 188       has_input_devices_(false), has_output_devices_(false) {
 189 }
 190
 191 MAYBE_WebRTCAudioDeviceTest::~MAYBE_WebRTCAudioDeviceTest() {}
 192
 193 void MAYBE_WebRTCAudioDeviceTest::SetUp() {
 194   // This part sets up a RenderThread environment to ensure that
 195   // RenderThread::current() (<=> TLS pointer) is valid.
 196   // Main parts are inspired by the RenderViewFakeResourcesTest.
 197   // Note that, the IPC part is not utilized in this test.
 198   saved_content_renderer_.reset(
 199       new ReplaceContentClientRenderer(&content_renderer_client_));
 200   mock_process_.reset(new WebRTCMockRenderProcess());
 201   ui_thread_.reset(
 202       new TestBrowserThread(BrowserThread::UI, base::MessageLoop::current()));
 203
 204   // Construct the resource context on the UI thread.
 205   resource_context_.reset(new MockRTCResourceContext);
 206
 207   static const char kThreadName[] = "RenderThread";
 208   ChildProcess::current()->io_message_loop()->PostTask(FROM_HERE,
 209       base::Bind(&MAYBE_WebRTCAudioDeviceTest::InitializeIOThread,
 210                  base::Unretained(this), kThreadName));
 211   WaitForIOThreadCompletion();
 212
 213   sandbox_was_enabled_ =
 214       RendererWebKitPlatformSupportImpl::SetSandboxEnabledForTesting(false);
 215   render_thread_ = new RenderThreadImpl(kThreadName);
 216 }
 217
 218 void MAYBE_WebRTCAudioDeviceTest::TearDown() {
 219   SetAudioHardwareConfig(NULL);
 220
 221   // Run any pending cleanup tasks that may have been posted to the main thread.
 222   base::RunLoop().RunUntilIdle();
 223
 224   // Kick of the cleanup process by closing the channel. This queues up
 225   // OnStreamClosed calls to be executed on the audio thread.
 226   ChildProcess::current()->io_message_loop()->PostTask(FROM_HERE,
 227       base::Bind(&MAYBE_WebRTCAudioDeviceTest::DestroyChannel,
 228                  base::Unretained(this)));
 229   WaitForIOThreadCompletion();
 230
 231   // When audio [input] render hosts are notified that the channel has
 232   // been closed, they post tasks to the audio thread to close the
 233   // AudioOutputController and once that's completed, a task is posted back to
 234   // the IO thread to actually delete the AudioEntry for the audio stream. Only
 235   // then is the reference to the audio manager released, so we wait for the
 236   // whole thing to be torn down before we finally uninitialize the io thread.
 237   WaitForAudioManagerCompletion();
 238
 239   ChildProcess::current()->io_message_loop()->PostTask(FROM_HERE,
 240       base::Bind(&MAYBE_WebRTCAudioDeviceTest::UninitializeIOThread,
 241                  base::Unretained((this))));
 242   WaitForIOThreadCompletion();
 243   mock_process_.reset();
 244   media_stream_manager_.reset();
 245   mirroring_manager_.reset();
 246   RendererWebKitPlatformSupportImpl::SetSandboxEnabledForTesting(
 247       sandbox_was_enabled_);
 248 }
 249
 250 bool MAYBE_WebRTCAudioDeviceTest::Send(IPC::Message* message) {
 251   return channel_->Send(message);
 252 }
 253
 254 void MAYBE_WebRTCAudioDeviceTest::SetAudioHardwareConfig(
 255     media::AudioHardwareConfig* hardware_config) {
 256   audio_hardware_config_ = hardware_config;
 257 }
 258
 259 scoped_refptr<WebRtcAudioRenderer>
 260 MAYBE_WebRTCAudioDeviceTest::CreateDefaultWebRtcAudioRenderer(
 261     int render_view_id) {
 262   media::AudioHardwareConfig* hardware_config =
 263       RenderThreadImpl::current()->GetAudioHardwareConfig();
 264   int sample_rate = hardware_config->GetOutputSampleRate();
 265   int frames_per_buffer = hardware_config->GetOutputBufferSize();
 266
 267   return new WebRtcAudioRenderer(render_view_id, 0, sample_rate,
 268                                  frames_per_buffer);
 269 }
 270
 271 void MAYBE_WebRTCAudioDeviceTest::InitializeIOThread(const char* thread_name) {
 272 #if defined(OS_WIN)
 273   // We initialize COM (STA) on our IO thread as is done in Chrome.
 274   // See BrowserProcessSubThread::Init.
 275   initialize_com_.reset(new base::win::ScopedCOMInitializer());
 276 #endif
 277
 278   // Set the current thread as the IO thread.
 279   io_thread_.reset(
 280       new TestBrowserThread(BrowserThread::IO, base::MessageLoop::current()));
 281
 282   // Populate our resource context.
 283   test_request_context_.reset(new net::TestURLRequestContext());
 284   MockRTCResourceContext* resource_context =
 285       static_cast<MockRTCResourceContext*>(resource_context_.get());
 286   resource_context->set_request_context(test_request_context_.get());
 287   media_internals_.reset(new MockMediaInternals());
 288
 289   // Create our own AudioManager, AudioMirroringManager and MediaStreamManager.
 290   audio_manager_.reset(media::AudioManager::Create());
 291   mirroring_manager_.reset(new AudioMirroringManager());
 292   media_stream_manager_.reset(new MediaStreamManager(audio_manager_.get()));
 293
 294   has_input_devices_ = audio_manager_->HasAudioInputDevices();
 295   has_output_devices_ = audio_manager_->HasAudioOutputDevices();
 296
 297   // Create an IPC channel that handles incoming messages on the IO thread.
 298   CreateChannel(thread_name);
 299 }
 300
 301 void MAYBE_WebRTCAudioDeviceTest::UninitializeIOThread() {
 302   resource_context_.reset();
 303
 304   test_request_context_.reset();
 305
 306 #if defined(OS_WIN)
 307   initialize_com_.reset();
 308 #endif
 309
 310   audio_manager_.reset();
 311 }
 312
 313 void MAYBE_WebRTCAudioDeviceTest::CreateChannel(const char* name) {
 314   DCHECK(BrowserThread::CurrentlyOn(BrowserThread::IO));
 315
 316   channel_.reset(new IPC::Channel(name, IPC::Channel::MODE_SERVER, this));
 317   ASSERT_TRUE(channel_->Connect());
 318
 319   static const int kRenderProcessId = 1;
 320   audio_render_host_ = new TestAudioRendererHost(
 321       kRenderProcessId, audio_manager_.get(), mirroring_manager_.get(),
 322       media_internals_.get(), media_stream_manager_.get(), channel_.get());
 323   audio_render_host_->set_peer_pid_for_testing(base::GetCurrentProcId());
 324
 325   audio_input_renderer_host_ =
 326       new TestAudioInputRendererHost(audio_manager_.get(),
 327                                      media_stream_manager_.get(),
 328                                      mirroring_manager_.get(),
 329                                      NULL,
 330                                      channel_.get());
 331   audio_input_renderer_host_->set_peer_pid_for_testing(
 332       base::GetCurrentProcId());
 333 }
 334
 335 void MAYBE_WebRTCAudioDeviceTest::DestroyChannel() {
 336   DCHECK(BrowserThread::CurrentlyOn(BrowserThread::IO));
 337   audio_render_host_->OnChannelClosing();
 338   audio_render_host_->OnFilterRemoved();
 339   audio_input_renderer_host_->OnChannelClosing();
 340   audio_input_renderer_host_->OnFilterRemoved();
 341   audio_render_host_->ResetChannel();
 342   audio_input_renderer_host_->ResetChannel();
 343   channel_.reset();
 344   audio_render_host_ = NULL;
 345   audio_input_renderer_host_ = NULL;
 346 }
 347
 348 void MAYBE_WebRTCAudioDeviceTest::OnGetAudioHardwareConfig(
 349     AudioParameters* input_params, AudioParameters* output_params) {
 350   ASSERT_TRUE(audio_hardware_config_);
 351   *input_params = audio_hardware_config_->GetInputConfig();
 352   *output_params = audio_hardware_config_->GetOutputConfig();
 353 }
 354
 355 // IPC::Listener implementation.
 356 bool MAYBE_WebRTCAudioDeviceTest::OnMessageReceived(
 357     const IPC::Message& message) {
 358   if (render_thread_) {
 359     IPC::ChannelProxy::MessageFilter* filter =
 360         render_thread_->audio_input_message_filter();
 361     if (filter->OnMessageReceived(message))
 362       return true;
 363
 364     filter = render_thread_->audio_message_filter();
 365     if (filter->OnMessageReceived(message))
 366       return true;
 367   }
 368
 369   if (audio_render_host_.get()) {
 370     bool message_was_ok = false;
 371     if (audio_render_host_->OnMessageReceived(message, &message_was_ok))
 372       return true;
 373   }
 374
 375   if (audio_input_renderer_host_.get()) {
 376     bool message_was_ok = false;
 377     if (audio_input_renderer_host_->OnMessageReceived(message, &message_was_ok))
 378       return true;
 379   }
 380
 381   bool handled ALLOW_UNUSED = true;
 382   bool message_is_ok = true;
 383   IPC_BEGIN_MESSAGE_MAP_EX(MAYBE_WebRTCAudioDeviceTest, message, message_is_ok)
 384     IPC_MESSAGE_HANDLER(ViewHostMsg_GetAudioHardwareConfig,
 385                         OnGetAudioHardwareConfig)
 386     IPC_MESSAGE_UNHANDLED(handled = false)
 387   IPC_END_MESSAGE_MAP_EX()
 388
 389   EXPECT_TRUE(message_is_ok);
 390
 391   return true;
 392 }
 393
 394 // Posts a final task to the IO message loop and waits for completion.
 395 void MAYBE_WebRTCAudioDeviceTest::WaitForIOThreadCompletion() {
 396   WaitForMessageLoopCompletion(
 397       ChildProcess::current()->io_message_loop()->message_loop_proxy().get());
 398 }
 399
 400 void MAYBE_WebRTCAudioDeviceTest::WaitForAudioManagerCompletion() {
 401   if (audio_manager_)
 402     WaitForMessageLoopCompletion(audio_manager_->GetMessageLoop().get());
 403 }
 404
 405 void MAYBE_WebRTCAudioDeviceTest::WaitForMessageLoopCompletion(
 406     base::MessageLoopProxy* loop) {
 407   base::WaitableEvent* event = new base::WaitableEvent(false, false);
 408   loop->PostTask(FROM_HERE, base::Bind(&base::WaitableEvent::Signal,
 409                  base::Unretained(event)));
 410   if (event->TimedWait(TestTimeouts::action_max_timeout())) {
 411     delete event;
 412   } else {
 413     // Don't delete the event object in case the message ever gets processed.
 414     // If we do, we will crash the test process.
 415     ADD_FAILURE() << "Failed to wait for message loop";
 416   }
 417 }
 418
 419 std::string MAYBE_WebRTCAudioDeviceTest::GetTestDataPath(
 420     const base::FilePath::StringType& file_name) {
 421   base::FilePath path;
 422   EXPECT_TRUE(PathService::Get(DIR_TEST_DATA, &path));
 423   path = path.Append(file_name);
 424   EXPECT_TRUE(base::PathExists(path));
 425 #if defined(OS_WIN)
 426   return WideToUTF8(path.value());
 427 #else
 428   return path.value();
 429 #endif
 430 }
 431
 432 WebRTCTransportImpl::WebRTCTransportImpl(webrtc::VoENetwork* network)
 433     : network_(network) {
 434 }
 435
 436 WebRTCTransportImpl::~WebRTCTransportImpl() {}
 437
 438 int WebRTCTransportImpl::SendPacket(int channel, const void* data, int len) {
 439   return network_->ReceivedRTPPacket(channel, data, len);
 440 }
 441
 442 int WebRTCTransportImpl::SendRTCPPacket(int channel, const void* data,
 443                                         int len) {
 444   return network_->ReceivedRTCPPacket(channel, data, len);
 445 }
 446
 447 }  // namespace content