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