src/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   CleanupPeerConnectionFactory();
 178   if (aec_dump_message_filter_.get())
 179     aec_dump_message_filter_->RemoveDelegate(this);
 180 }
 181
 182 blink::WebRTCPeerConnectionHandler*
 183 PeerConnectionDependencyFactory::CreateRTCPeerConnectionHandler(
 184     blink::WebRTCPeerConnectionHandlerClient* client) {
 185   // Save histogram data so we can see how much PeerConnetion is used.
 186   // The histogram counts the number of calls to the JS API
 187   // webKitRTCPeerConnection.
 188   UpdateWebRTCMethodCount(WEBKIT_RTC_PEER_CONNECTION);
 189
 190   return new RTCPeerConnectionHandler(client, this);
 191 }
 192
 193 bool PeerConnectionDependencyFactory::InitializeMediaStreamAudioSource(
 194     int render_view_id,
 195     const blink::WebMediaConstraints& audio_constraints,
 196     MediaStreamAudioSource* source_data) {
 197   DVLOG(1) << "InitializeMediaStreamAudioSources()";
 198
 199   // Do additional source initialization if the audio source is a valid
 200   // microphone or tab audio.
 201   RTCMediaConstraints native_audio_constraints(audio_constraints);
 202   MediaAudioConstraints::ApplyFixedAudioConstraints(&native_audio_constraints);
 203
 204   StreamDeviceInfo device_info = source_data->device_info();
 205   RTCMediaConstraints constraints = native_audio_constraints;
 206   // May modify both |constraints| and |effects|.
 207   HarmonizeConstraintsAndEffects(&constraints,
 208                                  &device_info.device.input.effects);
 209
 210   scoped_refptr<WebRtcAudioCapturer> capturer(
 211       CreateAudioCapturer(render_view_id, device_info, audio_constraints,
 212                           source_data));
 213   if (!capturer.get()) {
 214     const std::string log_string =
 215         "PCDF::InitializeMediaStreamAudioSource: fails to create capturer";
 216     WebRtcLogMessage(log_string);
 217     DVLOG(1) << log_string;
 218     // TODO(xians): Don't we need to check if source_observer is observing
 219     // something? If not, then it looks like we have a leak here.
 220     // OTOH, if it _is_ observing something, then the callback might
 221     // be called multiple times which is likely also a bug.
 222     return false;
 223   }
 224   source_data->SetAudioCapturer(capturer.get());
 225
 226   // Creates a LocalAudioSource object which holds audio options.
 227   // TODO(xians): The option should apply to the track instead of the source.
 228   // TODO(perkj): Move audio constraints parsing to Chrome.
 229   // Currently there are a few constraints that are parsed by libjingle and
 230   // the state is set to ended if parsing fails.
 231   scoped_refptr<webrtc::AudioSourceInterface> rtc_source(
 232       CreateLocalAudioSource(&constraints).get());
 233   if (rtc_source->state() != webrtc::MediaSourceInterface::kLive) {
 234     DLOG(WARNING) << "Failed to create rtc LocalAudioSource.";
 235     return false;
 236   }
 237   source_data->SetLocalAudioSource(rtc_source.get());
 238   return true;
 239 }
 240
 241 WebRtcVideoCapturerAdapter*
 242 PeerConnectionDependencyFactory::CreateVideoCapturer(
 243     bool is_screeencast) {
 244   // We need to make sure the libjingle thread wrappers have been created
 245   // before we can use an instance of a WebRtcVideoCapturerAdapter. This is
 246   // since the base class of WebRtcVideoCapturerAdapter is a
 247   // cricket::VideoCapturer and it uses the libjingle thread wrappers.
 248   if (!GetPcFactory().get())
 249     return NULL;
 250   return new WebRtcVideoCapturerAdapter(is_screeencast);
 251 }
 252
 253 scoped_refptr<webrtc::VideoSourceInterface>
 254 PeerConnectionDependencyFactory::CreateVideoSource(
 255     cricket::VideoCapturer* capturer,
 256     const blink::WebMediaConstraints& constraints) {
 257   RTCMediaConstraints webrtc_constraints(constraints);
 258   scoped_refptr<webrtc::VideoSourceInterface> source =
 259       GetPcFactory()->CreateVideoSource(capturer, &webrtc_constraints).get();
 260   return source;
 261 }
 262
 263 const scoped_refptr<webrtc::PeerConnectionFactoryInterface>&
 264 PeerConnectionDependencyFactory::GetPcFactory() {
 265   if (!pc_factory_.get())
 266     CreatePeerConnectionFactory();
 267   CHECK(pc_factory_.get());
 268   return pc_factory_;
 269 }
 270
 271 void PeerConnectionDependencyFactory::CreatePeerConnectionFactory() {
 272   DCHECK(!pc_factory_.get());
 273   DCHECK(!signaling_thread_);
 274   DCHECK(!worker_thread_);
 275   DCHECK(!network_manager_);
 276   DCHECK(!socket_factory_);
 277   DCHECK(!chrome_signaling_thread_.IsRunning());
 278   DCHECK(!chrome_worker_thread_.IsRunning());
 279
 280   DVLOG(1) << "PeerConnectionDependencyFactory::CreatePeerConnectionFactory()";
 281
 282   // To allow sending to the signaling/worker threads.
 283   jingle_glue::JingleThreadWrapper::EnsureForCurrentMessageLoop();
 284   jingle_glue::JingleThreadWrapper::current()->set_send_allowed(true);
 285
 286   CHECK(chrome_signaling_thread_.Start());
 287   CHECK(chrome_worker_thread_.Start());
 288
 289   base::WaitableEvent start_worker_event(true, false);
 290   chrome_worker_thread_.message_loop()->PostTask(FROM_HERE, base::Bind(
 291       &PeerConnectionDependencyFactory::InitializeWorkerThread,
 292       base::Unretained(this),
 293       &worker_thread_,
 294       &start_worker_event));
 295
 296   base::WaitableEvent create_network_manager_event(true, false);
 297   chrome_worker_thread_.message_loop()->PostTask(FROM_HERE, base::Bind(
 298       &PeerConnectionDependencyFactory::CreateIpcNetworkManagerOnWorkerThread,
 299       base::Unretained(this),
 300       &create_network_manager_event));
 301
 302   start_worker_event.Wait();
 303   create_network_manager_event.Wait();
 304
 305   CHECK(worker_thread_);
 306
 307   // Init SSL, which will be needed by PeerConnection.
 308 #if defined(USE_OPENSSL)
 309   if (!rtc::InitializeSSL()) {
 310     LOG(ERROR) << "Failed on InitializeSSL.";
 311     NOTREACHED();
 312     return;
 313   }
 314 #else
 315   // TODO(ronghuawu): Replace this call with InitializeSSL.
 316   net::EnsureNSSSSLInit();
 317 #endif
 318
 319   base::WaitableEvent start_signaling_event(true, false);
 320   chrome_signaling_thread_.message_loop()->PostTask(FROM_HERE, base::Bind(
 321       &PeerConnectionDependencyFactory::InitializeSignalingThread,
 322       base::Unretained(this),
 323       RenderThreadImpl::current()->GetGpuFactories(),
 324       &start_signaling_event));
 325
 326   // TODO(xians): Remove the following code after kDisableAudioTrackProcessing
 327   // is removed.
 328   if (!MediaStreamAudioProcessor::IsAudioTrackProcessingEnabled()) {
 329     aec_dump_message_filter_ = AecDumpMessageFilter::Get();
 330     // In unit tests not creating a message filter, |aec_dump_message_filter_|
 331     // will be NULL. We can just ignore that. Other unit tests and browser tests
 332     // ensure that we do get the filter when we should.
 333     if (aec_dump_message_filter_.get())
 334       aec_dump_message_filter_->AddDelegate(this);
 335   }
 336
 337   start_signaling_event.Wait();
 338   CHECK(signaling_thread_);
 339 }
 340
 341 void PeerConnectionDependencyFactory::InitializeSignalingThread(
 342     const scoped_refptr<media::GpuVideoAcceleratorFactories>& gpu_factories,
 343     base::WaitableEvent* event) {
 344   DCHECK(chrome_signaling_thread_.task_runner()->BelongsToCurrentThread());
 345   DCHECK(worker_thread_);
 346   DCHECK(p2p_socket_dispatcher_.get());
 347
 348   jingle_glue::JingleThreadWrapper::EnsureForCurrentMessageLoop();
 349   jingle_glue::JingleThreadWrapper::current()->set_send_allowed(true);
 350   signaling_thread_ = jingle_glue::JingleThreadWrapper::current();
 351
 352   EnsureWebRtcAudioDeviceImpl();
 353
 354   socket_factory_.reset(
 355       new IpcPacketSocketFactory(p2p_socket_dispatcher_.get()));
 356
 357   scoped_ptr<cricket::WebRtcVideoDecoderFactory> decoder_factory;
 358   scoped_ptr<cricket::WebRtcVideoEncoderFactory> encoder_factory;
 359
 360   const CommandLine* cmd_line = CommandLine::ForCurrentProcess();
 361   if (gpu_factories.get()) {
 362     if (!cmd_line->HasSwitch(switches::kDisableWebRtcHWDecoding))
 363       decoder_factory.reset(new RTCVideoDecoderFactory(gpu_factories));
 364
 365     if (!cmd_line->HasSwitch(switches::kDisableWebRtcHWEncoding))
 366       encoder_factory.reset(new RTCVideoEncoderFactory(gpu_factories));
 367   }
 368
 369 #if defined(OS_ANDROID)
 370   if (!media::MediaCodecBridge::SupportsSetParameters())
 371     encoder_factory.reset();
 372 #endif
 373
 374   pc_factory_ = webrtc::CreatePeerConnectionFactory(
 375       worker_thread_, signaling_thread_, audio_device_.get(),
 376       encoder_factory.release(), decoder_factory.release());
 377   CHECK(pc_factory_.get());
 378
 379   webrtc::PeerConnectionFactoryInterface::Options factory_options;
 380   factory_options.disable_sctp_data_channels = false;
 381   factory_options.disable_encryption =
 382       cmd_line->HasSwitch(switches::kDisableWebRtcEncryption);
 383   pc_factory_->SetOptions(factory_options);
 384
 385   event->Signal();
 386 }
 387
 388 bool PeerConnectionDependencyFactory::PeerConnectionFactoryCreated() {
 389   return pc_factory_.get() != NULL;
 390 }
 391
 392 scoped_refptr<webrtc::PeerConnectionInterface>
 393 PeerConnectionDependencyFactory::CreatePeerConnection(
 394     const webrtc::PeerConnectionInterface::RTCConfiguration& config,
 395     const webrtc::MediaConstraintsInterface* constraints,
 396     blink::WebFrame* web_frame,
 397     webrtc::PeerConnectionObserver* observer) {
 398   CHECK(web_frame);
 399   CHECK(observer);
 400   if (!GetPcFactory().get())
 401     return NULL;
 402
 403   scoped_refptr<P2PPortAllocatorFactory> pa_factory =
 404         new rtc::RefCountedObject<P2PPortAllocatorFactory>(
 405             p2p_socket_dispatcher_.get(),
 406             network_manager_,
 407             socket_factory_.get());
 408
 409   PeerConnectionIdentityService* identity_service =
 410       new PeerConnectionIdentityService(
 411           GURL(web_frame->document().url().spec()).GetOrigin());
 412
 413   return GetPcFactory()->CreatePeerConnection(config,
 414                                               constraints,
 415                                               pa_factory.get(),
 416                                               identity_service,
 417                                               observer).get();
 418 }
 419
 420 scoped_refptr<webrtc::MediaStreamInterface>
 421 PeerConnectionDependencyFactory::CreateLocalMediaStream(
 422     const std::string& label) {
 423   return GetPcFactory()->CreateLocalMediaStream(label).get();
 424 }
 425
 426 scoped_refptr<webrtc::AudioSourceInterface>
 427 PeerConnectionDependencyFactory::CreateLocalAudioSource(
 428     const webrtc::MediaConstraintsInterface* constraints) {
 429   scoped_refptr<webrtc::AudioSourceInterface> source =
 430       GetPcFactory()->CreateAudioSource(constraints).get();
 431   return source;
 432 }
 433
 434 void PeerConnectionDependencyFactory::CreateLocalAudioTrack(
 435     const blink::WebMediaStreamTrack& track) {
 436   blink::WebMediaStreamSource source = track.source();
 437   DCHECK_EQ(source.type(), blink::WebMediaStreamSource::TypeAudio);
 438   MediaStreamAudioSource* source_data =
 439       static_cast<MediaStreamAudioSource*>(source.extraData());
 440
 441   scoped_refptr<WebAudioCapturerSource> webaudio_source;
 442   if (!source_data) {
 443     if (source.requiresAudioConsumer()) {
 444       // We're adding a WebAudio MediaStream.
 445       // Create a specific capturer for each WebAudio consumer.
 446       webaudio_source = CreateWebAudioSource(&source);
 447       source_data =
 448           static_cast<MediaStreamAudioSource*>(source.extraData());
 449     } else {
 450       // TODO(perkj): Implement support for sources from
 451       // remote MediaStreams.
 452       NOTIMPLEMENTED();
 453       return;
 454     }
 455   }
 456
 457   // Creates an adapter to hold all the libjingle objects.
 458   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
 459       WebRtcLocalAudioTrackAdapter::Create(track.id().utf8(),
 460                                            source_data->local_audio_source()));
 461   static_cast<webrtc::AudioTrackInterface*>(adapter.get())->set_enabled(
 462       track.isEnabled());
 463
 464   // TODO(xians): Merge |source| to the capturer(). We can't do this today
 465   // because only one capturer() is supported while one |source| is created
 466   // for each audio track.
 467   scoped_ptr<WebRtcLocalAudioTrack> audio_track(new WebRtcLocalAudioTrack(
 468       adapter.get(), source_data->GetAudioCapturer(), webaudio_source.get()));
 469
 470   StartLocalAudioTrack(audio_track.get());
 471
 472   // Pass the ownership of the native local audio track to the blink track.
 473   blink::WebMediaStreamTrack writable_track = track;
 474   writable_track.setExtraData(audio_track.release());
 475 }
 476
 477 void PeerConnectionDependencyFactory::StartLocalAudioTrack(
 478     WebRtcLocalAudioTrack* audio_track) {
 479   // Add the WebRtcAudioDevice as the sink to the local audio track.
 480   // TODO(xians): Remove the following line of code after the APM in WebRTC is
 481   // completely deprecated. See http://crbug/365672.
 482   if (!MediaStreamAudioProcessor::IsAudioTrackProcessingEnabled())
 483     audio_track->AddSink(GetWebRtcAudioDevice());
 484
 485   // Start the audio track. This will hook the |audio_track| to the capturer
 486   // as the sink of the audio, and only start the source of the capturer if
 487   // it is the first audio track connecting to the capturer.
 488   audio_track->Start();
 489 }
 490
 491 scoped_refptr<WebAudioCapturerSource>
 492 PeerConnectionDependencyFactory::CreateWebAudioSource(
 493     blink::WebMediaStreamSource* source) {
 494   DVLOG(1) << "PeerConnectionDependencyFactory::CreateWebAudioSource()";
 495
 496   scoped_refptr<WebAudioCapturerSource>
 497       webaudio_capturer_source(new WebAudioCapturerSource());
 498   MediaStreamAudioSource* source_data = new MediaStreamAudioSource();
 499
 500   // Use the current default capturer for the WebAudio track so that the
 501   // WebAudio track can pass a valid delay value and |need_audio_processing|
 502   // flag to PeerConnection.
 503   // TODO(xians): Remove this after moving APM to Chrome.
 504   if (GetWebRtcAudioDevice()) {
 505     source_data->SetAudioCapturer(
 506         GetWebRtcAudioDevice()->GetDefaultCapturer());
 507   }
 508
 509   // Create a LocalAudioSource object which holds audio options.
 510   // SetLocalAudioSource() affects core audio parts in third_party/Libjingle.
 511   source_data->SetLocalAudioSource(CreateLocalAudioSource(NULL).get());
 512   source->setExtraData(source_data);
 513
 514   // Replace the default source with WebAudio as source instead.
 515   source->addAudioConsumer(webaudio_capturer_source.get());
 516
 517   return webaudio_capturer_source;
 518 }
 519
 520 scoped_refptr<webrtc::VideoTrackInterface>
 521 PeerConnectionDependencyFactory::CreateLocalVideoTrack(
 522     const std::string& id,
 523     webrtc::VideoSourceInterface* source) {
 524   return GetPcFactory()->CreateVideoTrack(id, source).get();
 525 }
 526
 527 scoped_refptr<webrtc::VideoTrackInterface>
 528 PeerConnectionDependencyFactory::CreateLocalVideoTrack(
 529     const std::string& id, cricket::VideoCapturer* capturer) {
 530   if (!capturer) {
 531     LOG(ERROR) << "CreateLocalVideoTrack called with null VideoCapturer.";
 532     return NULL;
 533   }
 534
 535   // Create video source from the |capturer|.
 536   scoped_refptr<webrtc::VideoSourceInterface> source =
 537       GetPcFactory()->CreateVideoSource(capturer, NULL).get();
 538
 539   // Create native track from the source.
 540   return GetPcFactory()->CreateVideoTrack(id, source.get()).get();
 541 }
 542
 543 webrtc::SessionDescriptionInterface*
 544 PeerConnectionDependencyFactory::CreateSessionDescription(
 545     const std::string& type,
 546     const std::string& sdp,
 547     webrtc::SdpParseError* error) {
 548   return webrtc::CreateSessionDescription(type, sdp, error);
 549 }
 550
 551 webrtc::IceCandidateInterface*
 552 PeerConnectionDependencyFactory::CreateIceCandidate(
 553     const std::string& sdp_mid,
 554     int sdp_mline_index,
 555     const std::string& sdp) {
 556   return webrtc::CreateIceCandidate(sdp_mid, sdp_mline_index, sdp);
 557 }
 558
 559 WebRtcAudioDeviceImpl*
 560 PeerConnectionDependencyFactory::GetWebRtcAudioDevice() {
 561   return audio_device_.get();
 562 }
 563
 564 void PeerConnectionDependencyFactory::InitializeWorkerThread(
 565     rtc::Thread** thread,
 566     base::WaitableEvent* event) {
 567   jingle_glue::JingleThreadWrapper::EnsureForCurrentMessageLoop();
 568   jingle_glue::JingleThreadWrapper::current()->set_send_allowed(true);
 569   *thread = jingle_glue::JingleThreadWrapper::current();
 570   event->Signal();
 571 }
 572
 573 void PeerConnectionDependencyFactory::CreateIpcNetworkManagerOnWorkerThread(
 574     base::WaitableEvent* event) {
 575   DCHECK_EQ(base::MessageLoop::current(), chrome_worker_thread_.message_loop());
 576   network_manager_ = new IpcNetworkManager(p2p_socket_dispatcher_.get());
 577   event->Signal();
 578 }
 579
 580 void PeerConnectionDependencyFactory::DeleteIpcNetworkManager() {
 581   DCHECK_EQ(base::MessageLoop::current(), chrome_worker_thread_.message_loop());
 582   delete network_manager_;
 583   network_manager_ = NULL;
 584 }
 585
 586 void PeerConnectionDependencyFactory::CleanupPeerConnectionFactory() {
 587   pc_factory_ = NULL;
 588   if (network_manager_) {
 589     // The network manager needs to free its resources on the thread they were
 590     // created, which is the worked thread.
 591     if (chrome_worker_thread_.IsRunning()) {
 592       chrome_worker_thread_.message_loop()->PostTask(FROM_HERE, base::Bind(
 593           &PeerConnectionDependencyFactory::DeleteIpcNetworkManager,
 594           base::Unretained(this)));
 595       // Stopping the thread will wait until all tasks have been
 596       // processed before returning. We wait for the above task to finish before
 597       // letting the the function continue to avoid any potential race issues.
 598       chrome_worker_thread_.Stop();
 599     } else {
 600       NOTREACHED() << "Worker thread not running.";
 601     }
 602   }
 603 }
 604
 605 scoped_refptr<WebRtcAudioCapturer>
 606 PeerConnectionDependencyFactory::CreateAudioCapturer(
 607     int render_view_id,
 608     const StreamDeviceInfo& device_info,
 609     const blink::WebMediaConstraints& constraints,
 610     MediaStreamAudioSource* audio_source) {
 611   // TODO(xians): Handle the cases when gUM is called without a proper render
 612   // view, for example, by an extension.
 613   DCHECK_GE(render_view_id, 0);
 614
 615   EnsureWebRtcAudioDeviceImpl();
 616   DCHECK(GetWebRtcAudioDevice());
 617   return WebRtcAudioCapturer::CreateCapturer(render_view_id, device_info,
 618                                              constraints,
 619                                              GetWebRtcAudioDevice(),
 620                                              audio_source);
 621 }
 622
 623 scoped_refptr<base::MessageLoopProxy>
 624 PeerConnectionDependencyFactory::GetWebRtcWorkerThread() const {
 625   DCHECK(CalledOnValidThread());
 626   return chrome_worker_thread_.message_loop_proxy();
 627 }
 628
 629 scoped_refptr<base::MessageLoopProxy>
 630 PeerConnectionDependencyFactory::GetWebRtcSignalingThread() const {
 631   DCHECK(CalledOnValidThread());
 632   return chrome_signaling_thread_.message_loop_proxy();
 633 }
 634
 635 void PeerConnectionDependencyFactory::OnAecDumpFile(
 636     const IPC::PlatformFileForTransit& file_handle) {
 637   DCHECK(CalledOnValidThread());
 638   DCHECK(!MediaStreamAudioProcessor::IsAudioTrackProcessingEnabled());
 639   DCHECK(PeerConnectionFactoryCreated());
 640
 641   base::File file = IPC::PlatformFileForTransitToFile(file_handle);
 642   DCHECK(file.IsValid());
 643
 644   // |pc_factory_| always takes ownership of |aec_dump_file|. If StartAecDump()
 645   // fails, |aec_dump_file| will be closed.
 646   if (!GetPcFactory()->StartAecDump(file.TakePlatformFile()))
 647     VLOG(1) << "Could not start AEC dump.";
 648 }
 649
 650 void PeerConnectionDependencyFactory::OnDisableAecDump() {
 651   DCHECK(CalledOnValidThread());
 652   DCHECK(!MediaStreamAudioProcessor::IsAudioTrackProcessingEnabled());
 653   // Do nothing. We never disable AEC dump for non-track-processing case.
 654 }
 655
 656 void PeerConnectionDependencyFactory::OnIpcClosing() {
 657   DCHECK(CalledOnValidThread());
 658   aec_dump_message_filter_ = NULL;
 659 }
 660
 661 void PeerConnectionDependencyFactory::EnsureWebRtcAudioDeviceImpl() {
 662   if (audio_device_.get())
 663     return;
 664
 665   audio_device_ = new WebRtcAudioDeviceImpl();
 666 }
 667
 668 }  // namespace content