content/renderer/media/webrtc/peer_connection_dependency_factory.cc

   1 // Copyright 2014 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/renderer/media/webrtc/peer_connection_dependency_factory.h"
   6
   7 #include <vector>
   8
   9 #include "base/command_line.h"
  10 #include "base/strings/utf_string_conversions.h"
  11 #include "base/synchronization/waitable_event.h"
  12 #include "content/common/media/media_stream_messages.h"
  13 #include "content/public/common/content_switches.h"
  14 #include "content/public/common/renderer_preferences.h"
  15 #include "content/renderer/media/media_stream.h"
  16 #include "content/renderer/media/media_stream_audio_processor.h"
  17 #include "content/renderer/media/media_stream_audio_processor_options.h"
  18 #include "content/renderer/media/media_stream_audio_source.h"
  19 #include "content/renderer/media/media_stream_video_source.h"
  20 #include "content/renderer/media/media_stream_video_track.h"
  21 #include "content/renderer/media/peer_connection_identity_service.h"
  22 #include "content/renderer/media/rtc_media_constraints.h"
  23 #include "content/renderer/media/rtc_peer_connection_handler.h"
  24 #include "content/renderer/media/rtc_video_decoder_factory.h"
  25 #include "content/renderer/media/rtc_video_encoder_factory.h"
  26 #include "content/renderer/media/webaudio_capturer_source.h"
  27 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
  28 #include "content/renderer/media/webrtc/webrtc_video_capturer_adapter.h"
  29 #include "content/renderer/media/webrtc_audio_device_impl.h"
  30 #include "content/renderer/media/webrtc_local_audio_track.h"
  31 #include "content/renderer/media/webrtc_logging.h"
  32 #include "content/renderer/media/webrtc_uma_histograms.h"
  33 #include "content/renderer/p2p/ipc_network_manager.h"
  34 #include "content/renderer/p2p/ipc_socket_factory.h"
  35 #include "content/renderer/p2p/port_allocator.h"
  36 #include "content/renderer/render_thread_impl.h"
  37 #include "content/renderer/render_view_impl.h"
  38 #include "jingle/glue/thread_wrapper.h"
  39 #include "media/renderers/gpu_video_accelerator_factories.h"
  40 #include "third_party/WebKit/public/platform/WebMediaConstraints.h"
  41 #include "third_party/WebKit/public/platform/WebMediaStream.h"
  42 #include "third_party/WebKit/public/platform/WebMediaStreamSource.h"
  43 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h"
  44 #include "third_party/WebKit/public/platform/WebURL.h"
  45 #include "third_party/WebKit/public/web/WebDocument.h"
  46 #include "third_party/WebKit/public/web/WebFrame.h"
  47 #include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface.h"
  48
  49 #if defined(USE_OPENSSL)
  50 #include "third_party/webrtc/base/ssladapter.h"
  51 #else
  52 #include "net/socket/nss_ssl_util.h"
  53 #endif
  54
  55 #if defined(OS_ANDROID)
  56 #include "media/base/android/media_codec_bridge.h"
  57 #endif
  58
  59 namespace content {
  60
  61 // Map of corresponding media constraints and platform effects.
  62 struct {
  63   const char* constraint;
  64   const media::AudioParameters::PlatformEffectsMask effect;
  65 } const kConstraintEffectMap[] = {
  66   { content::kMediaStreamAudioDucking,
  67     media::AudioParameters::DUCKING },
  68   { webrtc::MediaConstraintsInterface::kEchoCancellation,
  69     media::AudioParameters::ECHO_CANCELLER },
  70 };
  71
  72 // If any platform effects are available, check them against the constraints.
  73 // Disable effects to match false constraints, but if a constraint is true, set
  74 // the constraint to false to later disable the software effect.
  75 //
  76 // This function may modify both |constraints| and |effects|.
  77 void HarmonizeConstraintsAndEffects(RTCMediaConstraints* constraints,
  78                                     int* effects) {
  79   if (*effects != media::AudioParameters::NO_EFFECTS) {
  80     for (size_t i = 0; i < arraysize(kConstraintEffectMap); ++i) {
  81       bool value;
  82       size_t is_mandatory = 0;
  83       if (!webrtc::FindConstraint(constraints,
  84                                   kConstraintEffectMap[i].constraint,
  85                                   &value,
  86                                   &is_mandatory) || !value) {
  87         // If the constraint is false, or does not exist, disable the platform
  88         // effect.
  89         *effects &= ~kConstraintEffectMap[i].effect;
  90         DVLOG(1) << "Disabling platform effect: "
  91                  << kConstraintEffectMap[i].effect;
  92       } else if (*effects & kConstraintEffectMap[i].effect) {
  93         // If the constraint is true, leave the platform effect enabled, and
  94         // set the constraint to false to later disable the software effect.
  95         if (is_mandatory) {
  96           constraints->AddMandatory(kConstraintEffectMap[i].constraint,
  97               webrtc::MediaConstraintsInterface::kValueFalse, true);
  98         } else {
  99           constraints->AddOptional(kConstraintEffectMap[i].constraint,
 100               webrtc::MediaConstraintsInterface::kValueFalse, true);
 101         }
 102         DVLOG(1) << "Disabling constraint: "
 103                  << kConstraintEffectMap[i].constraint;
 104       } else if (kConstraintEffectMap[i].effect ==
 105                  media::AudioParameters::DUCKING && value && !is_mandatory) {
 106         // Special handling of the DUCKING flag that sets the optional
 107         // constraint to |false| to match what the device will support.
 108         constraints->AddOptional(kConstraintEffectMap[i].constraint,
 109             webrtc::MediaConstraintsInterface::kValueFalse, true);
 110         // No need to modify |effects| since the ducking flag is already off.
 111         DCHECK((*effects & media::AudioParameters::DUCKING) == 0);
 112       }
 113     }
 114   }
 115 }
 116
 117 class P2PPortAllocatorFactory : public webrtc::PortAllocatorFactoryInterface {
 118  public:
 119   P2PPortAllocatorFactory(P2PSocketDispatcher* socket_dispatcher,
 120                           rtc::NetworkManager* network_manager,
 121                           rtc::PacketSocketFactory* socket_factory,
 122                           const GURL& origin,
 123                           bool enable_multiple_routes)
 124       : socket_dispatcher_(socket_dispatcher),
 125         network_manager_(network_manager),
 126         socket_factory_(socket_factory),
 127         origin_(origin),
 128         enable_multiple_routes_(enable_multiple_routes) {}
 129
 130   cricket::PortAllocator* CreatePortAllocator(
 131       const std::vector<StunConfiguration>& stun_servers,
 132       const std::vector<TurnConfiguration>& turn_configurations) override {
 133     P2PPortAllocator::Config config;
 134     for (size_t i = 0; i < stun_servers.size(); ++i) {
 135       config.stun_servers.insert(rtc::SocketAddress(
 136           stun_servers[i].server.hostname(),
 137           stun_servers[i].server.port()));
 138     }
 139     for (size_t i = 0; i < turn_configurations.size(); ++i) {
 140       P2PPortAllocator::Config::RelayServerConfig relay_config;
 141       relay_config.server_address = turn_configurations[i].server.hostname();
 142       relay_config.port = turn_configurations[i].server.port();
 143       relay_config.username = turn_configurations[i].username;
 144       relay_config.password = turn_configurations[i].password;
 145       relay_config.transport_type = turn_configurations[i].transport_type;
 146       relay_config.secure = turn_configurations[i].secure;
 147       config.relays.push_back(relay_config);
 148
 149       // Use turn servers as stun servers.
 150       config.stun_servers.insert(rtc::SocketAddress(
 151           turn_configurations[i].server.hostname(),
 152           turn_configurations[i].server.port()));
 153     }
 154     config.enable_multiple_routes = enable_multiple_routes_;
 155
 156     return new P2PPortAllocator(
 157         socket_dispatcher_.get(), network_manager_,
 158         socket_factory_, config, origin_);
 159   }
 160
 161  protected:
 162   ~P2PPortAllocatorFactory() override {}
 163
 164  private:
 165   scoped_refptr<P2PSocketDispatcher> socket_dispatcher_;
 166   // |network_manager_| and |socket_factory_| are a weak references, owned by
 167   // PeerConnectionDependencyFactory.
 168   rtc::NetworkManager* network_manager_;
 169   rtc::PacketSocketFactory* socket_factory_;
 170   // The origin URL of the WebFrame that created the
 171   // P2PPortAllocatorFactory.
 172   GURL origin_;
 173   // When false, only 'any' address (all 0s) will be bound for address
 174   // discovery.
 175   bool enable_multiple_routes_;
 176 };
 177
 178 PeerConnectionDependencyFactory::PeerConnectionDependencyFactory(
 179     P2PSocketDispatcher* p2p_socket_dispatcher)
 180     : network_manager_(NULL),
 181       p2p_socket_dispatcher_(p2p_socket_dispatcher),
 182       signaling_thread_(NULL),
 183       worker_thread_(NULL),
 184       chrome_signaling_thread_("Chrome_libJingle_Signaling"),
 185       chrome_worker_thread_("Chrome_libJingle_WorkerThread") {
 186 }
 187
 188 PeerConnectionDependencyFactory::~PeerConnectionDependencyFactory() {
 189   DVLOG(1) << "~PeerConnectionDependencyFactory()";
 190   DCHECK(pc_factory_ == NULL);
 191 }
 192
 193 blink::WebRTCPeerConnectionHandler*
 194 PeerConnectionDependencyFactory::CreateRTCPeerConnectionHandler(
 195     blink::WebRTCPeerConnectionHandlerClient* client) {
 196   // Save histogram data so we can see how much PeerConnetion is used.
 197   // The histogram counts the number of calls to the JS API
 198   // webKitRTCPeerConnection.
 199   UpdateWebRTCMethodCount(WEBKIT_RTC_PEER_CONNECTION);
 200
 201   return new RTCPeerConnectionHandler(client, this);
 202 }
 203
 204 bool PeerConnectionDependencyFactory::InitializeMediaStreamAudioSource(
 205     int render_frame_id,
 206     const blink::WebMediaConstraints& audio_constraints,
 207     MediaStreamAudioSource* source_data) {
 208   DVLOG(1) << "InitializeMediaStreamAudioSources()";
 209
 210   // Do additional source initialization if the audio source is a valid
 211   // microphone or tab audio.
 212   RTCMediaConstraints native_audio_constraints(audio_constraints);
 213   MediaAudioConstraints::ApplyFixedAudioConstraints(&native_audio_constraints);
 214
 215   StreamDeviceInfo device_info = source_data->device_info();
 216   RTCMediaConstraints constraints = native_audio_constraints;
 217   // May modify both |constraints| and |effects|.
 218   HarmonizeConstraintsAndEffects(&constraints,
 219                                  &device_info.device.input.effects);
 220
 221   scoped_refptr<WebRtcAudioCapturer> capturer(CreateAudioCapturer(
 222       render_frame_id, device_info, audio_constraints, source_data));
 223   if (!capturer.get()) {
 224     const std::string log_string =
 225         "PCDF::InitializeMediaStreamAudioSource: fails to create capturer";
 226     WebRtcLogMessage(log_string);
 227     DVLOG(1) << log_string;
 228     // TODO(xians): Don't we need to check if source_observer is observing
 229     // something? If not, then it looks like we have a leak here.
 230     // OTOH, if it _is_ observing something, then the callback might
 231     // be called multiple times which is likely also a bug.
 232     return false;
 233   }
 234   source_data->SetAudioCapturer(capturer.get());
 235
 236   // Creates a LocalAudioSource object which holds audio options.
 237   // TODO(xians): The option should apply to the track instead of the source.
 238   // TODO(perkj): Move audio constraints parsing to Chrome.
 239   // Currently there are a few constraints that are parsed by libjingle and
 240   // the state is set to ended if parsing fails.
 241   scoped_refptr<webrtc::AudioSourceInterface> rtc_source(
 242       CreateLocalAudioSource(&constraints).get());
 243   if (rtc_source->state() != webrtc::MediaSourceInterface::kLive) {
 244     DLOG(WARNING) << "Failed to create rtc LocalAudioSource.";
 245     return false;
 246   }
 247   source_data->SetLocalAudioSource(rtc_source.get());
 248   return true;
 249 }
 250
 251 WebRtcVideoCapturerAdapter*
 252 PeerConnectionDependencyFactory::CreateVideoCapturer(
 253     bool is_screeencast) {
 254   // We need to make sure the libjingle thread wrappers have been created
 255   // before we can use an instance of a WebRtcVideoCapturerAdapter. This is
 256   // since the base class of WebRtcVideoCapturerAdapter is a
 257   // cricket::VideoCapturer and it uses the libjingle thread wrappers.
 258   if (!GetPcFactory().get())
 259     return NULL;
 260   return new WebRtcVideoCapturerAdapter(is_screeencast);
 261 }
 262
 263 scoped_refptr<webrtc::VideoSourceInterface>
 264 PeerConnectionDependencyFactory::CreateVideoSource(
 265     cricket::VideoCapturer* capturer,
 266     const blink::WebMediaConstraints& constraints) {
 267   RTCMediaConstraints webrtc_constraints(constraints);
 268   scoped_refptr<webrtc::VideoSourceInterface> source =
 269       GetPcFactory()->CreateVideoSource(capturer, &webrtc_constraints).get();
 270   return source;
 271 }
 272
 273 const scoped_refptr<webrtc::PeerConnectionFactoryInterface>&
 274 PeerConnectionDependencyFactory::GetPcFactory() {
 275   if (!pc_factory_.get())
 276     CreatePeerConnectionFactory();
 277   CHECK(pc_factory_.get());
 278   return pc_factory_;
 279 }
 280
 281
 282 void PeerConnectionDependencyFactory::WillDestroyCurrentMessageLoop() {
 283   CleanupPeerConnectionFactory();
 284 }
 285
 286 void PeerConnectionDependencyFactory::CreatePeerConnectionFactory() {
 287   DCHECK(!pc_factory_.get());
 288   DCHECK(!signaling_thread_);
 289   DCHECK(!worker_thread_);
 290   DCHECK(!network_manager_);
 291   DCHECK(!socket_factory_);
 292   DCHECK(!chrome_signaling_thread_.IsRunning());
 293   DCHECK(!chrome_worker_thread_.IsRunning());
 294
 295   DVLOG(1) << "PeerConnectionDependencyFactory::CreatePeerConnectionFactory()";
 296
 297   base::MessageLoop::current()->AddDestructionObserver(this);
 298   // To allow sending to the signaling/worker threads.
 299   jingle_glue::JingleThreadWrapper::EnsureForCurrentMessageLoop();
 300   jingle_glue::JingleThreadWrapper::current()->set_send_allowed(true);
 301
 302   CHECK(chrome_signaling_thread_.Start());
 303   CHECK(chrome_worker_thread_.Start());
 304
 305   base::WaitableEvent start_worker_event(true, false);
 306   chrome_worker_thread_.message_loop()->PostTask(FROM_HERE, base::Bind(
 307       &PeerConnectionDependencyFactory::InitializeWorkerThread,
 308       base::Unretained(this),
 309       &worker_thread_,
 310       &start_worker_event));
 311
 312   base::WaitableEvent create_network_manager_event(true, false);
 313   chrome_worker_thread_.message_loop()->PostTask(FROM_HERE, base::Bind(
 314       &PeerConnectionDependencyFactory::CreateIpcNetworkManagerOnWorkerThread,
 315       base::Unretained(this),
 316       &create_network_manager_event));
 317
 318   start_worker_event.Wait();
 319   create_network_manager_event.Wait();
 320
 321   CHECK(worker_thread_);
 322
 323   // Init SSL, which will be needed by PeerConnection.
 324 #if defined(USE_OPENSSL)
 325   if (!rtc::InitializeSSL()) {
 326     LOG(ERROR) << "Failed on InitializeSSL.";
 327     NOTREACHED();
 328     return;
 329   }
 330 #else
 331   // TODO(ronghuawu): Replace this call with InitializeSSL.
 332   net::EnsureNSSSSLInit();
 333 #endif
 334
 335   base::WaitableEvent start_signaling_event(true, false);
 336   chrome_signaling_thread_.message_loop()->PostTask(FROM_HERE, base::Bind(
 337       &PeerConnectionDependencyFactory::InitializeSignalingThread,
 338       base::Unretained(this),
 339       RenderThreadImpl::current()->GetGpuFactories(),
 340       &start_signaling_event));
 341
 342   start_signaling_event.Wait();
 343   CHECK(signaling_thread_);
 344 }
 345
 346 void PeerConnectionDependencyFactory::InitializeSignalingThread(
 347     const scoped_refptr<media::GpuVideoAcceleratorFactories>& gpu_factories,
 348     base::WaitableEvent* event) {
 349   DCHECK(chrome_signaling_thread_.task_runner()->BelongsToCurrentThread());
 350   DCHECK(worker_thread_);
 351   DCHECK(p2p_socket_dispatcher_.get());
 352
 353   jingle_glue::JingleThreadWrapper::EnsureForCurrentMessageLoop();
 354   jingle_glue::JingleThreadWrapper::current()->set_send_allowed(true);
 355   signaling_thread_ = jingle_glue::JingleThreadWrapper::current();
 356
 357   EnsureWebRtcAudioDeviceImpl();
 358
 359   socket_factory_.reset(
 360       new IpcPacketSocketFactory(p2p_socket_dispatcher_.get()));
 361
 362   scoped_ptr<cricket::WebRtcVideoDecoderFactory> decoder_factory;
 363   scoped_ptr<cricket::WebRtcVideoEncoderFactory> encoder_factory;
 364
 365   const base::CommandLine* cmd_line = base::CommandLine::ForCurrentProcess();
 366   if (gpu_factories.get()) {
 367     if (!cmd_line->HasSwitch(switches::kDisableWebRtcHWDecoding))
 368       decoder_factory.reset(new RTCVideoDecoderFactory(gpu_factories));
 369
 370     if (!cmd_line->HasSwitch(switches::kDisableWebRtcHWEncoding))
 371       encoder_factory.reset(new RTCVideoEncoderFactory(gpu_factories));
 372   }
 373
 374 #if defined(OS_ANDROID)
 375   if (!media::MediaCodecBridge::SupportsSetParameters())
 376     encoder_factory.reset();
 377 #endif
 378
 379   pc_factory_ = webrtc::CreatePeerConnectionFactory(
 380       worker_thread_, signaling_thread_, audio_device_.get(),
 381       encoder_factory.release(), decoder_factory.release());
 382   CHECK(pc_factory_.get());
 383
 384   webrtc::PeerConnectionFactoryInterface::Options factory_options;
 385   factory_options.disable_sctp_data_channels = false;
 386   factory_options.disable_encryption =
 387       cmd_line->HasSwitch(switches::kDisableWebRtcEncryption);
 388   pc_factory_->SetOptions(factory_options);
 389
 390   event->Signal();
 391 }
 392
 393 bool PeerConnectionDependencyFactory::PeerConnectionFactoryCreated() {
 394   return pc_factory_.get() != NULL;
 395 }
 396
 397 scoped_refptr<webrtc::PeerConnectionInterface>
 398 PeerConnectionDependencyFactory::CreatePeerConnection(
 399     const webrtc::PeerConnectionInterface::RTCConfiguration& config,
 400     const webrtc::MediaConstraintsInterface* constraints,
 401     blink::WebFrame* web_frame,
 402     webrtc::PeerConnectionObserver* observer) {
 403   CHECK(web_frame);
 404   CHECK(observer);
 405   if (!GetPcFactory().get())
 406     return NULL;
 407
 408   // Copy the flag from Preference associated with this WebFrame.
 409   bool enable_multiple_routes = true;
 410   if (web_frame && web_frame->view()) {
 411     RenderViewImpl* renderer_view_impl =
 412         RenderViewImpl::FromWebView(web_frame->view());
 413     if (renderer_view_impl) {
 414       enable_multiple_routes = renderer_view_impl->renderer_preferences()
 415                                     .enable_webrtc_multiple_routes;
 416     }
 417   }
 418
 419   scoped_refptr<P2PPortAllocatorFactory> pa_factory =
 420       new rtc::RefCountedObject<P2PPortAllocatorFactory>(
 421           p2p_socket_dispatcher_.get(), network_manager_, socket_factory_.get(),
 422           GURL(web_frame->document().url().spec()).GetOrigin(),
 423           enable_multiple_routes);
 424
 425   PeerConnectionIdentityService* identity_service =
 426       new PeerConnectionIdentityService(
 427           GURL(web_frame->document().url().spec()).GetOrigin());
 428
 429   return GetPcFactory()->CreatePeerConnection(config,
 430                                               constraints,
 431                                               pa_factory.get(),
 432                                               identity_service,
 433                                               observer).get();
 434 }
 435
 436 scoped_refptr<webrtc::MediaStreamInterface>
 437 PeerConnectionDependencyFactory::CreateLocalMediaStream(
 438     const std::string& label) {
 439   return GetPcFactory()->CreateLocalMediaStream(label).get();
 440 }
 441
 442 scoped_refptr<webrtc::AudioSourceInterface>
 443 PeerConnectionDependencyFactory::CreateLocalAudioSource(
 444     const webrtc::MediaConstraintsInterface* constraints) {
 445   scoped_refptr<webrtc::AudioSourceInterface> source =
 446       GetPcFactory()->CreateAudioSource(constraints).get();
 447   return source;
 448 }
 449
 450 void PeerConnectionDependencyFactory::CreateLocalAudioTrack(
 451     const blink::WebMediaStreamTrack& track) {
 452   blink::WebMediaStreamSource source = track.source();
 453   DCHECK_EQ(source.type(), blink::WebMediaStreamSource::TypeAudio);
 454   MediaStreamAudioSource* source_data =
 455       static_cast<MediaStreamAudioSource*>(source.extraData());
 456
 457   scoped_refptr<WebAudioCapturerSource> webaudio_source;
 458   if (!source_data) {
 459     if (source.requiresAudioConsumer()) {
 460       // We're adding a WebAudio MediaStream.
 461       // Create a specific capturer for each WebAudio consumer.
 462       webaudio_source = CreateWebAudioSource(&source);
 463       source_data =
 464           static_cast<MediaStreamAudioSource*>(source.extraData());
 465     } else {
 466       // TODO(perkj): Implement support for sources from
 467       // remote MediaStreams.
 468       NOTIMPLEMENTED();
 469       return;
 470     }
 471   }
 472
 473   // Creates an adapter to hold all the libjingle objects.
 474   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
 475       WebRtcLocalAudioTrackAdapter::Create(track.id().utf8(),
 476                                            source_data->local_audio_source()));
 477   static_cast<webrtc::AudioTrackInterface*>(adapter.get())->set_enabled(
 478       track.isEnabled());
 479
 480   // TODO(xians): Merge |source| to the capturer(). We can't do this today
 481   // because only one capturer() is supported while one |source| is created
 482   // for each audio track.
 483   scoped_ptr<WebRtcLocalAudioTrack> audio_track(new WebRtcLocalAudioTrack(
 484       adapter.get(), source_data->GetAudioCapturer(), webaudio_source.get()));
 485
 486   StartLocalAudioTrack(audio_track.get());
 487
 488   // Pass the ownership of the native local audio track to the blink track.
 489   blink::WebMediaStreamTrack writable_track = track;
 490   writable_track.setExtraData(audio_track.release());
 491 }
 492
 493 void PeerConnectionDependencyFactory::StartLocalAudioTrack(
 494     WebRtcLocalAudioTrack* audio_track) {
 495   // Start the audio track. This will hook the |audio_track| to the capturer
 496   // as the sink of the audio, and only start the source of the capturer if
 497   // it is the first audio track connecting to the capturer.
 498   audio_track->Start();
 499 }
 500
 501 scoped_refptr<WebAudioCapturerSource>
 502 PeerConnectionDependencyFactory::CreateWebAudioSource(
 503     blink::WebMediaStreamSource* source) {
 504   DVLOG(1) << "PeerConnectionDependencyFactory::CreateWebAudioSource()";
 505
 506   scoped_refptr<WebAudioCapturerSource>
 507       webaudio_capturer_source(new WebAudioCapturerSource());
 508   MediaStreamAudioSource* source_data = new MediaStreamAudioSource();
 509
 510   // Use the current default capturer for the WebAudio track so that the
 511   // WebAudio track can pass a valid delay value and |need_audio_processing|
 512   // flag to PeerConnection.
 513   // TODO(xians): Remove this after moving APM to Chrome.
 514   if (GetWebRtcAudioDevice()) {
 515     source_data->SetAudioCapturer(
 516         GetWebRtcAudioDevice()->GetDefaultCapturer());
 517   }
 518
 519   // Create a LocalAudioSource object which holds audio options.
 520   // SetLocalAudioSource() affects core audio parts in third_party/Libjingle.
 521   source_data->SetLocalAudioSource(CreateLocalAudioSource(NULL).get());
 522   source->setExtraData(source_data);
 523
 524   // Replace the default source with WebAudio as source instead.
 525   source->addAudioConsumer(webaudio_capturer_source.get());
 526
 527   return webaudio_capturer_source;
 528 }
 529
 530 scoped_refptr<webrtc::VideoTrackInterface>
 531 PeerConnectionDependencyFactory::CreateLocalVideoTrack(
 532     const std::string& id,
 533     webrtc::VideoSourceInterface* source) {
 534   return GetPcFactory()->CreateVideoTrack(id, source).get();
 535 }
 536
 537 scoped_refptr<webrtc::VideoTrackInterface>
 538 PeerConnectionDependencyFactory::CreateLocalVideoTrack(
 539     const std::string& id, cricket::VideoCapturer* capturer) {
 540   if (!capturer) {
 541     LOG(ERROR) << "CreateLocalVideoTrack called with null VideoCapturer.";
 542     return NULL;
 543   }
 544
 545   // Create video source from the |capturer|.
 546   scoped_refptr<webrtc::VideoSourceInterface> source =
 547       GetPcFactory()->CreateVideoSource(capturer, NULL).get();
 548
 549   // Create native track from the source.
 550   return GetPcFactory()->CreateVideoTrack(id, source.get()).get();
 551 }
 552
 553 webrtc::SessionDescriptionInterface*
 554 PeerConnectionDependencyFactory::CreateSessionDescription(
 555     const std::string& type,
 556     const std::string& sdp,
 557     webrtc::SdpParseError* error) {
 558   return webrtc::CreateSessionDescription(type, sdp, error);
 559 }
 560
 561 webrtc::IceCandidateInterface*
 562 PeerConnectionDependencyFactory::CreateIceCandidate(
 563     const std::string& sdp_mid,
 564     int sdp_mline_index,
 565     const std::string& sdp) {
 566   return webrtc::CreateIceCandidate(sdp_mid, sdp_mline_index, sdp);
 567 }
 568
 569 WebRtcAudioDeviceImpl*
 570 PeerConnectionDependencyFactory::GetWebRtcAudioDevice() {
 571   return audio_device_.get();
 572 }
 573
 574 void PeerConnectionDependencyFactory::InitializeWorkerThread(
 575     rtc::Thread** thread,
 576     base::WaitableEvent* event) {
 577   jingle_glue::JingleThreadWrapper::EnsureForCurrentMessageLoop();
 578   jingle_glue::JingleThreadWrapper::current()->set_send_allowed(true);
 579   *thread = jingle_glue::JingleThreadWrapper::current();
 580   event->Signal();
 581 }
 582
 583 void PeerConnectionDependencyFactory::CreateIpcNetworkManagerOnWorkerThread(
 584     base::WaitableEvent* event) {
 585   DCHECK_EQ(base::MessageLoop::current(), chrome_worker_thread_.message_loop());
 586   network_manager_ = new IpcNetworkManager(p2p_socket_dispatcher_.get());
 587   event->Signal();
 588 }
 589
 590 void PeerConnectionDependencyFactory::DeleteIpcNetworkManager() {
 591   DCHECK_EQ(base::MessageLoop::current(), chrome_worker_thread_.message_loop());
 592   delete network_manager_;
 593   network_manager_ = NULL;
 594 }
 595
 596 void PeerConnectionDependencyFactory::CleanupPeerConnectionFactory() {
 597   DVLOG(1) << "PeerConnectionDependencyFactory::CleanupPeerConnectionFactory()";
 598   pc_factory_ = NULL;
 599   if (network_manager_) {
 600     // The network manager needs to free its resources on the thread they were
 601     // created, which is the worked thread.
 602     if (chrome_worker_thread_.IsRunning()) {
 603       chrome_worker_thread_.message_loop()->PostTask(FROM_HERE, base::Bind(
 604           &PeerConnectionDependencyFactory::DeleteIpcNetworkManager,
 605           base::Unretained(this)));
 606       // Stopping the thread will wait until all tasks have been
 607       // processed before returning. We wait for the above task to finish before
 608       // letting the the function continue to avoid any potential race issues.
 609       chrome_worker_thread_.Stop();
 610     } else {
 611       NOTREACHED() << "Worker thread not running.";
 612     }
 613   }
 614 }
 615
 616 scoped_refptr<WebRtcAudioCapturer>
 617 PeerConnectionDependencyFactory::CreateAudioCapturer(
 618     int render_frame_id,
 619     const StreamDeviceInfo& device_info,
 620     const blink::WebMediaConstraints& constraints,
 621     MediaStreamAudioSource* audio_source) {
 622   // TODO(xians): Handle the cases when gUM is called without a proper render
 623   // view, for example, by an extension.
 624   DCHECK_GE(render_frame_id, 0);
 625
 626   EnsureWebRtcAudioDeviceImpl();
 627   DCHECK(GetWebRtcAudioDevice());
 628   return WebRtcAudioCapturer::CreateCapturer(
 629       render_frame_id, device_info, constraints, GetWebRtcAudioDevice(),
 630       audio_source);
 631 }
 632
 633 scoped_refptr<base::MessageLoopProxy>
 634 PeerConnectionDependencyFactory::GetWebRtcWorkerThread() const {
 635   DCHECK(CalledOnValidThread());
 636   return chrome_worker_thread_.message_loop_proxy();
 637 }
 638
 639 scoped_refptr<base::MessageLoopProxy>
 640 PeerConnectionDependencyFactory::GetWebRtcSignalingThread() const {
 641   DCHECK(CalledOnValidThread());
 642   return chrome_signaling_thread_.message_loop_proxy();
 643 }
 644
 645 void PeerConnectionDependencyFactory::EnsureWebRtcAudioDeviceImpl() {
 646   if (audio_device_.get())
 647     return;
 648
 649   audio_device_ = new WebRtcAudioDeviceImpl();
 650 }
 651
 652 }  // namespace content