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