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_uma_histograms.h"
  31 #include "content/renderer/p2p/ipc_network_manager.h"
  32 #include "content/renderer/p2p/ipc_socket_factory.h"
  33 #include "content/renderer/p2p/port_allocator.h"
  34 #include "content/renderer/render_thread_impl.h"
  35 #include "jingle/glue/thread_wrapper.h"
  36 #include "media/filters/gpu_video_accelerator_factories.h"
  37 #include "third_party/WebKit/public/platform/WebMediaConstraints.h"
  38 #include "third_party/WebKit/public/platform/WebMediaStream.h"
  39 #include "third_party/WebKit/public/platform/WebMediaStreamSource.h"
  40 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h"
  41 #include "third_party/WebKit/public/platform/WebURL.h"
  42 #include "third_party/WebKit/public/web/WebDocument.h"
  43 #include "third_party/WebKit/public/web/WebFrame.h"
  44 #include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface.h"
  45
  46 #if defined(USE_OPENSSL)
  47 #include "third_party/webrtc/base/ssladapter.h"
  48 #else
  49 #include "net/socket/nss_ssl_util.h"
  50 #endif
  51
  52 #if defined(OS_ANDROID)
  53 #include "media/base/android/media_codec_bridge.h"
  54 #endif
  55
  56 namespace content {
  57
  58 // Map of corresponding media constraints and platform effects.
  59 struct {
  60   const char* constraint;
  61   const media::AudioParameters::PlatformEffectsMask effect;
  62 } const kConstraintEffectMap[] = {
  63   { content::kMediaStreamAudioDucking,
  64     media::AudioParameters::DUCKING },
  65   { webrtc::MediaConstraintsInterface::kEchoCancellation,
  66     media::AudioParameters::ECHO_CANCELLER },
  67 };
  68
  69 // If any platform effects are available, check them against the constraints.
  70 // Disable effects to match false constraints, but if a constraint is true, set
  71 // the constraint to false to later disable the software effect.
  72 //
  73 // This function may modify both |constraints| and |effects|.
  74 void HarmonizeConstraintsAndEffects(RTCMediaConstraints* constraints,
  75                                     int* effects) {
  76   if (*effects != media::AudioParameters::NO_EFFECTS) {
  77     for (size_t i = 0; i < ARRAYSIZE_UNSAFE(kConstraintEffectMap); ++i) {
  78       bool value;
  79       size_t is_mandatory = 0;
  80       if (!webrtc::FindConstraint(constraints,
  81                                   kConstraintEffectMap[i].constraint,
  82                                   &value,
  83                                   &is_mandatory) || !value) {
  84         // If the constraint is false, or does not exist, disable the platform
  85         // effect.
  86         *effects &= ~kConstraintEffectMap[i].effect;
  87         DVLOG(1) << "Disabling platform effect: "
  88                  << kConstraintEffectMap[i].effect;
  89       } else if (*effects & kConstraintEffectMap[i].effect) {
  90         // If the constraint is true, leave the platform effect enabled, and
  91         // set the constraint to false to later disable the software effect.
  92         if (is_mandatory) {
  93           constraints->AddMandatory(kConstraintEffectMap[i].constraint,
  94               webrtc::MediaConstraintsInterface::kValueFalse, true);
  95         } else {
  96           constraints->AddOptional(kConstraintEffectMap[i].constraint,
  97               webrtc::MediaConstraintsInterface::kValueFalse, true);
  98         }
  99         DVLOG(1) << "Disabling constraint: "
 100                  << kConstraintEffectMap[i].constraint;
 101       } else if (kConstraintEffectMap[i].effect ==
 102                  media::AudioParameters::DUCKING && value && !is_mandatory) {
 103         // Special handling of the DUCKING flag that sets the optional
 104         // constraint to |false| to match what the device will support.
 105         constraints->AddOptional(kConstraintEffectMap[i].constraint,
 106             webrtc::MediaConstraintsInterface::kValueFalse, true);
 107         // No need to modify |effects| since the ducking flag is already off.
 108         DCHECK((*effects & media::AudioParameters::DUCKING) == 0);
 109       }
 110     }
 111   }
 112 }
 113
 114 class P2PPortAllocatorFactory : public webrtc::PortAllocatorFactoryInterface {
 115  public:
 116   P2PPortAllocatorFactory(
 117       P2PSocketDispatcher* socket_dispatcher,
 118       rtc::NetworkManager* network_manager,
 119       rtc::PacketSocketFactory* socket_factory,
 120       blink::WebFrame* web_frame)
 121       : socket_dispatcher_(socket_dispatcher),
 122         network_manager_(network_manager),
 123         socket_factory_(socket_factory),
 124         web_frame_(web_frame) {
 125   }
 126
 127   virtual cricket::PortAllocator* CreatePortAllocator(
 128       const std::vector<StunConfiguration>& stun_servers,
 129       const std::vector<TurnConfiguration>& turn_configurations) OVERRIDE {
 130     CHECK(web_frame_);
 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     config.legacy_relay = false;
 138     for (size_t i = 0; i < turn_configurations.size(); ++i) {
 139       P2PPortAllocator::Config::RelayServerConfig relay_config;
 140       relay_config.server_address = turn_configurations[i].server.hostname();
 141       relay_config.port = turn_configurations[i].server.port();
 142       relay_config.username = turn_configurations[i].username;
 143       relay_config.password = turn_configurations[i].password;
 144       relay_config.transport_type = turn_configurations[i].transport_type;
 145       relay_config.secure = turn_configurations[i].secure;
 146       config.relays.push_back(relay_config);
 147
 148       // Use turn servers as stun servers.
 149       config.stun_servers.insert(rtc::SocketAddress(
 150           turn_configurations[i].server.hostname(),
 151           turn_configurations[i].server.port()));
 152     }
 153
 154     return new P2PPortAllocator(
 155         web_frame_, socket_dispatcher_.get(), network_manager_,
 156         socket_factory_, config);
 157   }
 158
 159  protected:
 160   virtual ~P2PPortAllocatorFactory() {}
 161
 162  private:
 163   scoped_refptr<P2PSocketDispatcher> socket_dispatcher_;
 164   // |network_manager_| and |socket_factory_| are a weak references, owned by
 165   // PeerConnectionDependencyFactory.
 166   rtc::NetworkManager* network_manager_;
 167   rtc::PacketSocketFactory* socket_factory_;
 168   // Raw ptr to the WebFrame that created the P2PPortAllocatorFactory.
 169   blink::WebFrame* web_frame_;
 170 };
 171
 172 PeerConnectionDependencyFactory::PeerConnectionDependencyFactory(
 173     P2PSocketDispatcher* p2p_socket_dispatcher)
 174     : network_manager_(NULL),
 175       p2p_socket_dispatcher_(p2p_socket_dispatcher),
 176       signaling_thread_(NULL),
 177       worker_thread_(NULL),
 178       chrome_worker_thread_("Chrome_libJingle_WorkerThread") {
 179 }
 180
 181 PeerConnectionDependencyFactory::~PeerConnectionDependencyFactory() {
 182   CleanupPeerConnectionFactory();
 183   if (aec_dump_message_filter_.get())
 184     aec_dump_message_filter_->RemoveDelegate(this);
 185 }
 186
 187 blink::WebRTCPeerConnectionHandler*
 188 PeerConnectionDependencyFactory::CreateRTCPeerConnectionHandler(
 189     blink::WebRTCPeerConnectionHandlerClient* client) {
 190   // Save histogram data so we can see how much PeerConnetion is used.
 191   // The histogram counts the number of calls to the JS API
 192   // webKitRTCPeerConnection.
 193   UpdateWebRTCMethodCount(WEBKIT_RTC_PEER_CONNECTION);
 194
 195   return new RTCPeerConnectionHandler(client, this);
 196 }
 197
 198 bool PeerConnectionDependencyFactory::InitializeMediaStreamAudioSource(
 199     int render_view_id,
 200     const blink::WebMediaConstraints& audio_constraints,
 201     MediaStreamAudioSource* source_data) {
 202   DVLOG(1) << "InitializeMediaStreamAudioSources()";
 203
 204   // Do additional source initialization if the audio source is a valid
 205   // microphone or tab audio.
 206   RTCMediaConstraints native_audio_constraints(audio_constraints);
 207   MediaAudioConstraints::ApplyFixedAudioConstraints(&native_audio_constraints);
 208
 209   StreamDeviceInfo device_info = source_data->device_info();
 210   RTCMediaConstraints constraints = native_audio_constraints;
 211   // May modify both |constraints| and |effects|.
 212   HarmonizeConstraintsAndEffects(&constraints,
 213                                  &device_info.device.input.effects);
 214
 215   scoped_refptr<WebRtcAudioCapturer> capturer(
 216       CreateAudioCapturer(render_view_id, device_info, audio_constraints,
 217                           source_data));
 218   if (!capturer.get()) {
 219     DLOG(WARNING) << "Failed to create the capturer for device "
 220         << device_info.device.id;
 221     // TODO(xians): Don't we need to check if source_observer is observing
 222     // something? If not, then it looks like we have a leak here.
 223     // OTOH, if it _is_ observing something, then the callback might
 224     // be called multiple times which is likely also a bug.
 225     return false;
 226   }
 227   source_data->SetAudioCapturer(capturer.get());
 228
 229   // Creates a LocalAudioSource object which holds audio options.
 230   // TODO(xians): The option should apply to the track instead of the source.
 231   // TODO(perkj): Move audio constraints parsing to Chrome.
 232   // Currently there are a few constraints that are parsed by libjingle and
 233   // the state is set to ended if parsing fails.
 234   scoped_refptr<webrtc::AudioSourceInterface> rtc_source(
 235       CreateLocalAudioSource(&constraints).get());
 236   if (rtc_source->state() != webrtc::MediaSourceInterface::kLive) {
 237     DLOG(WARNING) << "Failed to create rtc LocalAudioSource.";
 238     return false;
 239   }
 240   source_data->SetLocalAudioSource(rtc_source.get());
 241   return true;
 242 }
 243
 244 WebRtcVideoCapturerAdapter*
 245 PeerConnectionDependencyFactory::CreateVideoCapturer(
 246     bool is_screeencast) {
 247   // We need to make sure the libjingle thread wrappers have been created
 248   // before we can use an instance of a WebRtcVideoCapturerAdapter. This is
 249   // since the base class of WebRtcVideoCapturerAdapter is a
 250   // cricket::VideoCapturer and it uses the libjingle thread wrappers.
 251   if (!GetPcFactory().get())
 252     return NULL;
 253   return new WebRtcVideoCapturerAdapter(is_screeencast);
 254 }
 255
 256 scoped_refptr<webrtc::VideoSourceInterface>
 257 PeerConnectionDependencyFactory::CreateVideoSource(
 258     cricket::VideoCapturer* capturer,
 259     const blink::WebMediaConstraints& constraints) {
 260   RTCMediaConstraints webrtc_constraints(constraints);
 261   scoped_refptr<webrtc::VideoSourceInterface> source =
 262       GetPcFactory()->CreateVideoSource(capturer, &webrtc_constraints).get();
 263   return source;
 264 }
 265
 266 const scoped_refptr<webrtc::PeerConnectionFactoryInterface>&
 267 PeerConnectionDependencyFactory::GetPcFactory() {
 268   if (!pc_factory_.get())
 269     CreatePeerConnectionFactory();
 270   CHECK(pc_factory_.get());
 271   return pc_factory_;
 272 }
 273
 274 void PeerConnectionDependencyFactory::CreatePeerConnectionFactory() {
 275   DCHECK(!pc_factory_.get());
 276   DCHECK(!signaling_thread_);
 277   DCHECK(!worker_thread_);
 278   DCHECK(!network_manager_);
 279   DCHECK(!socket_factory_);
 280   DCHECK(!chrome_worker_thread_.IsRunning());
 281
 282   DVLOG(1) << "PeerConnectionDependencyFactory::CreatePeerConnectionFactory()";
 283
 284   jingle_glue::JingleThreadWrapper::EnsureForCurrentMessageLoop();
 285   jingle_glue::JingleThreadWrapper::current()->set_send_allowed(true);
 286   signaling_thread_ = jingle_glue::JingleThreadWrapper::current();
 287   CHECK(signaling_thread_);
 288
 289   CHECK(chrome_worker_thread_.Start());
 290
 291   base::WaitableEvent start_worker_event(true, false);
 292   chrome_worker_thread_.message_loop()->PostTask(FROM_HERE, base::Bind(
 293       &PeerConnectionDependencyFactory::InitializeWorkerThread,
 294       base::Unretained(this),
 295       &worker_thread_,
 296       &start_worker_event));
 297   start_worker_event.Wait();
 298   CHECK(worker_thread_);
 299
 300   base::WaitableEvent create_network_manager_event(true, false);
 301   chrome_worker_thread_.message_loop()->PostTask(FROM_HERE, base::Bind(
 302       &PeerConnectionDependencyFactory::CreateIpcNetworkManagerOnWorkerThread,
 303       base::Unretained(this),
 304       &create_network_manager_event));
 305   create_network_manager_event.Wait();
 306
 307   socket_factory_.reset(
 308       new IpcPacketSocketFactory(p2p_socket_dispatcher_.get()));
 309
 310   // Init SSL, which will be needed by PeerConnection.
 311 #if defined(USE_OPENSSL)
 312   if (!rtc::InitializeSSL()) {
 313     LOG(ERROR) << "Failed on InitializeSSL.";
 314     NOTREACHED();
 315     return;
 316   }
 317 #else
 318   // TODO(ronghuawu): Replace this call with InitializeSSL.
 319   net::EnsureNSSSSLInit();
 320 #endif
 321
 322   scoped_ptr<cricket::WebRtcVideoDecoderFactory> decoder_factory;
 323   scoped_ptr<cricket::WebRtcVideoEncoderFactory> encoder_factory;
 324
 325   const CommandLine* cmd_line = CommandLine::ForCurrentProcess();
 326   scoped_refptr<media::GpuVideoAcceleratorFactories> gpu_factories =
 327       RenderThreadImpl::current()->GetGpuFactories();
 328   if (!cmd_line->HasSwitch(switches::kDisableWebRtcHWDecoding)) {
 329     if (gpu_factories.get())
 330       decoder_factory.reset(new RTCVideoDecoderFactory(gpu_factories));
 331   }
 332
 333   if (!cmd_line->HasSwitch(switches::kDisableWebRtcHWEncoding)) {
 334     if (gpu_factories.get())
 335       encoder_factory.reset(new RTCVideoEncoderFactory(gpu_factories));
 336   }
 337
 338 #if defined(OS_ANDROID)
 339   if (!media::MediaCodecBridge::SupportsSetParameters())
 340     encoder_factory.reset();
 341 #endif
 342
 343   EnsureWebRtcAudioDeviceImpl();
 344
 345   scoped_refptr<webrtc::PeerConnectionFactoryInterface> factory(
 346       webrtc::CreatePeerConnectionFactory(worker_thread_,
 347                                           signaling_thread_,
 348                                           audio_device_.get(),
 349                                           encoder_factory.release(),
 350                                           decoder_factory.release()));
 351   CHECK(factory.get());
 352
 353   pc_factory_ = factory;
 354   webrtc::PeerConnectionFactoryInterface::Options factory_options;
 355   factory_options.disable_sctp_data_channels = false;
 356   factory_options.disable_encryption =
 357       cmd_line->HasSwitch(switches::kDisableWebRtcEncryption);
 358   pc_factory_->SetOptions(factory_options);
 359
 360   // TODO(xians): Remove the following code after kDisableAudioTrackProcessing
 361   // is removed.
 362   if (!MediaStreamAudioProcessor::IsAudioTrackProcessingEnabled()) {
 363     aec_dump_message_filter_ = AecDumpMessageFilter::Get();
 364     // In unit tests not creating a message filter, |aec_dump_message_filter_|
 365     // will be NULL. We can just ignore that. Other unit tests and browser tests
 366     // ensure that we do get the filter when we should.
 367     if (aec_dump_message_filter_.get())
 368       aec_dump_message_filter_->AddDelegate(this);
 369   }
 370 }
 371
 372 bool PeerConnectionDependencyFactory::PeerConnectionFactoryCreated() {
 373   return pc_factory_.get() != NULL;
 374 }
 375
 376 scoped_refptr<webrtc::PeerConnectionInterface>
 377 PeerConnectionDependencyFactory::CreatePeerConnection(
 378     const webrtc::PeerConnectionInterface::RTCConfiguration& config,
 379     const webrtc::MediaConstraintsInterface* constraints,
 380     blink::WebFrame* web_frame,
 381     webrtc::PeerConnectionObserver* observer) {
 382   CHECK(web_frame);
 383   CHECK(observer);
 384   if (!GetPcFactory().get())
 385     return NULL;
 386
 387   scoped_refptr<P2PPortAllocatorFactory> pa_factory =
 388         new rtc::RefCountedObject<P2PPortAllocatorFactory>(
 389             p2p_socket_dispatcher_.get(),
 390             network_manager_,
 391             socket_factory_.get(),
 392             web_frame);
 393
 394   PeerConnectionIdentityService* identity_service =
 395       new PeerConnectionIdentityService(
 396           GURL(web_frame->document().url().spec()).GetOrigin());
 397
 398   return GetPcFactory()->CreatePeerConnection(config,
 399                                               constraints,
 400                                               pa_factory.get(),
 401                                               identity_service,
 402                                               observer).get();
 403 }
 404
 405 scoped_refptr<webrtc::MediaStreamInterface>
 406 PeerConnectionDependencyFactory::CreateLocalMediaStream(
 407     const std::string& label) {
 408   return GetPcFactory()->CreateLocalMediaStream(label).get();
 409 }
 410
 411 scoped_refptr<webrtc::AudioSourceInterface>
 412 PeerConnectionDependencyFactory::CreateLocalAudioSource(
 413     const webrtc::MediaConstraintsInterface* constraints) {
 414   scoped_refptr<webrtc::AudioSourceInterface> source =
 415       GetPcFactory()->CreateAudioSource(constraints).get();
 416   return source;
 417 }
 418
 419 void PeerConnectionDependencyFactory::CreateLocalAudioTrack(
 420     const blink::WebMediaStreamTrack& track) {
 421   blink::WebMediaStreamSource source = track.source();
 422   DCHECK_EQ(source.type(), blink::WebMediaStreamSource::TypeAudio);
 423   MediaStreamAudioSource* source_data =
 424       static_cast<MediaStreamAudioSource*>(source.extraData());
 425
 426   scoped_refptr<WebAudioCapturerSource> webaudio_source;
 427   if (!source_data) {
 428     if (source.requiresAudioConsumer()) {
 429       // We're adding a WebAudio MediaStream.
 430       // Create a specific capturer for each WebAudio consumer.
 431       webaudio_source = CreateWebAudioSource(&source);
 432       source_data =
 433           static_cast<MediaStreamAudioSource*>(source.extraData());
 434     } else {
 435       // TODO(perkj): Implement support for sources from
 436       // remote MediaStreams.
 437       NOTIMPLEMENTED();
 438       return;
 439     }
 440   }
 441
 442   // Creates an adapter to hold all the libjingle objects.
 443   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
 444       WebRtcLocalAudioTrackAdapter::Create(track.id().utf8(),
 445                                            source_data->local_audio_source()));
 446   static_cast<webrtc::AudioTrackInterface*>(adapter.get())->set_enabled(
 447       track.isEnabled());
 448
 449   // TODO(xians): Merge |source| to the capturer(). We can't do this today
 450   // because only one capturer() is supported while one |source| is created
 451   // for each audio track.
 452   scoped_ptr<WebRtcLocalAudioTrack> audio_track(new WebRtcLocalAudioTrack(
 453       adapter.get(), source_data->GetAudioCapturer(), webaudio_source.get()));
 454
 455   StartLocalAudioTrack(audio_track.get());
 456
 457   // Pass the ownership of the native local audio track to the blink track.
 458   blink::WebMediaStreamTrack writable_track = track;
 459   writable_track.setExtraData(audio_track.release());
 460 }
 461
 462 void PeerConnectionDependencyFactory::StartLocalAudioTrack(
 463     WebRtcLocalAudioTrack* audio_track) {
 464   // Add the WebRtcAudioDevice as the sink to the local audio track.
 465   // TODO(xians): Remove the following line of code after the APM in WebRTC is
 466   // completely deprecated. See http://crbug/365672.
 467   if (!MediaStreamAudioProcessor::IsAudioTrackProcessingEnabled())
 468     audio_track->AddSink(GetWebRtcAudioDevice());
 469
 470   // Start the audio track. This will hook the |audio_track| to the capturer
 471   // as the sink of the audio, and only start the source of the capturer if
 472   // it is the first audio track connecting to the capturer.
 473   audio_track->Start();
 474 }
 475
 476 scoped_refptr<WebAudioCapturerSource>
 477 PeerConnectionDependencyFactory::CreateWebAudioSource(
 478     blink::WebMediaStreamSource* source) {
 479   DVLOG(1) << "PeerConnectionDependencyFactory::CreateWebAudioSource()";
 480
 481   scoped_refptr<WebAudioCapturerSource>
 482       webaudio_capturer_source(new WebAudioCapturerSource());
 483   MediaStreamAudioSource* source_data = new MediaStreamAudioSource();
 484
 485   // Use the current default capturer for the WebAudio track so that the
 486   // WebAudio track can pass a valid delay value and |need_audio_processing|
 487   // flag to PeerConnection.
 488   // TODO(xians): Remove this after moving APM to Chrome.
 489   if (GetWebRtcAudioDevice()) {
 490     source_data->SetAudioCapturer(
 491         GetWebRtcAudioDevice()->GetDefaultCapturer());
 492   }
 493
 494   // Create a LocalAudioSource object which holds audio options.
 495   // SetLocalAudioSource() affects core audio parts in third_party/Libjingle.
 496   source_data->SetLocalAudioSource(CreateLocalAudioSource(NULL).get());
 497   source->setExtraData(source_data);
 498
 499   // Replace the default source with WebAudio as source instead.
 500   source->addAudioConsumer(webaudio_capturer_source.get());
 501
 502   return webaudio_capturer_source;
 503 }
 504
 505 scoped_refptr<webrtc::VideoTrackInterface>
 506 PeerConnectionDependencyFactory::CreateLocalVideoTrack(
 507     const std::string& id,
 508     webrtc::VideoSourceInterface* source) {
 509   return GetPcFactory()->CreateVideoTrack(id, source).get();
 510 }
 511
 512 scoped_refptr<webrtc::VideoTrackInterface>
 513 PeerConnectionDependencyFactory::CreateLocalVideoTrack(
 514     const std::string& id, cricket::VideoCapturer* capturer) {
 515   if (!capturer) {
 516     LOG(ERROR) << "CreateLocalVideoTrack called with null VideoCapturer.";
 517     return NULL;
 518   }
 519
 520   // Create video source from the |capturer|.
 521   scoped_refptr<webrtc::VideoSourceInterface> source =
 522       GetPcFactory()->CreateVideoSource(capturer, NULL).get();
 523
 524   // Create native track from the source.
 525   return GetPcFactory()->CreateVideoTrack(id, source.get()).get();
 526 }
 527
 528 webrtc::SessionDescriptionInterface*
 529 PeerConnectionDependencyFactory::CreateSessionDescription(
 530     const std::string& type,
 531     const std::string& sdp,
 532     webrtc::SdpParseError* error) {
 533   return webrtc::CreateSessionDescription(type, sdp, error);
 534 }
 535
 536 webrtc::IceCandidateInterface*
 537 PeerConnectionDependencyFactory::CreateIceCandidate(
 538     const std::string& sdp_mid,
 539     int sdp_mline_index,
 540     const std::string& sdp) {
 541   return webrtc::CreateIceCandidate(sdp_mid, sdp_mline_index, sdp);
 542 }
 543
 544 WebRtcAudioDeviceImpl*
 545 PeerConnectionDependencyFactory::GetWebRtcAudioDevice() {
 546   return audio_device_.get();
 547 }
 548
 549 void PeerConnectionDependencyFactory::InitializeWorkerThread(
 550     rtc::Thread** thread,
 551     base::WaitableEvent* event) {
 552   jingle_glue::JingleThreadWrapper::EnsureForCurrentMessageLoop();
 553   jingle_glue::JingleThreadWrapper::current()->set_send_allowed(true);
 554   *thread = jingle_glue::JingleThreadWrapper::current();
 555   event->Signal();
 556 }
 557
 558 void PeerConnectionDependencyFactory::CreateIpcNetworkManagerOnWorkerThread(
 559     base::WaitableEvent* event) {
 560   DCHECK_EQ(base::MessageLoop::current(), chrome_worker_thread_.message_loop());
 561   network_manager_ = new IpcNetworkManager(p2p_socket_dispatcher_.get());
 562   event->Signal();
 563 }
 564
 565 void PeerConnectionDependencyFactory::DeleteIpcNetworkManager() {
 566   DCHECK_EQ(base::MessageLoop::current(), chrome_worker_thread_.message_loop());
 567   delete network_manager_;
 568   network_manager_ = NULL;
 569 }
 570
 571 void PeerConnectionDependencyFactory::CleanupPeerConnectionFactory() {
 572   pc_factory_ = NULL;
 573   if (network_manager_) {
 574     // The network manager needs to free its resources on the thread they were
 575     // created, which is the worked thread.
 576     if (chrome_worker_thread_.IsRunning()) {
 577       chrome_worker_thread_.message_loop()->PostTask(FROM_HERE, base::Bind(
 578           &PeerConnectionDependencyFactory::DeleteIpcNetworkManager,
 579           base::Unretained(this)));
 580       // Stopping the thread will wait until all tasks have been
 581       // processed before returning. We wait for the above task to finish before
 582       // letting the the function continue to avoid any potential race issues.
 583       chrome_worker_thread_.Stop();
 584     } else {
 585       NOTREACHED() << "Worker thread not running.";
 586     }
 587   }
 588 }
 589
 590 scoped_refptr<WebRtcAudioCapturer>
 591 PeerConnectionDependencyFactory::CreateAudioCapturer(
 592     int render_view_id,
 593     const StreamDeviceInfo& device_info,
 594     const blink::WebMediaConstraints& constraints,
 595     MediaStreamAudioSource* audio_source) {
 596   // TODO(xians): Handle the cases when gUM is called without a proper render
 597   // view, for example, by an extension.
 598   DCHECK_GE(render_view_id, 0);
 599
 600   EnsureWebRtcAudioDeviceImpl();
 601   DCHECK(GetWebRtcAudioDevice());
 602   return WebRtcAudioCapturer::CreateCapturer(render_view_id, device_info,
 603                                              constraints,
 604                                              GetWebRtcAudioDevice(),
 605                                              audio_source);
 606 }
 607
 608 void PeerConnectionDependencyFactory::AddNativeAudioTrackToBlinkTrack(
 609     webrtc::MediaStreamTrackInterface* native_track,
 610     const blink::WebMediaStreamTrack& webkit_track,
 611     bool is_local_track) {
 612   DCHECK(!webkit_track.isNull() && !webkit_track.extraData());
 613   DCHECK_EQ(blink::WebMediaStreamSource::TypeAudio,
 614             webkit_track.source().type());
 615   blink::WebMediaStreamTrack track = webkit_track;
 616
 617   DVLOG(1) << "AddNativeTrackToBlinkTrack() audio";
 618   track.setExtraData(
 619       new MediaStreamTrack(
 620           static_cast<webrtc::AudioTrackInterface*>(native_track),
 621           is_local_track));
 622 }
 623
 624 scoped_refptr<base::MessageLoopProxy>
 625 PeerConnectionDependencyFactory::GetWebRtcWorkerThread() const {
 626   DCHECK(CalledOnValidThread());
 627   return chrome_worker_thread_.message_loop_proxy();
 628 }
 629
 630 void PeerConnectionDependencyFactory::OnAecDumpFile(
 631     const IPC::PlatformFileForTransit& file_handle) {
 632   DCHECK(CalledOnValidThread());
 633   DCHECK(!MediaStreamAudioProcessor::IsAudioTrackProcessingEnabled());
 634   DCHECK(PeerConnectionFactoryCreated());
 635
 636   base::File file = IPC::PlatformFileForTransitToFile(file_handle);
 637   DCHECK(file.IsValid());
 638
 639   // |pc_factory_| always takes ownership of |aec_dump_file|. If StartAecDump()
 640   // fails, |aec_dump_file| will be closed.
 641   if (!GetPcFactory()->StartAecDump(file.TakePlatformFile()))
 642     VLOG(1) << "Could not start AEC dump.";
 643 }
 644
 645 void PeerConnectionDependencyFactory::OnDisableAecDump() {
 646   DCHECK(CalledOnValidThread());
 647   DCHECK(!MediaStreamAudioProcessor::IsAudioTrackProcessingEnabled());
 648   // Do nothing. We never disable AEC dump for non-track-processing case.
 649 }
 650
 651 void PeerConnectionDependencyFactory::OnIpcClosing() {
 652   DCHECK(CalledOnValidThread());
 653   aec_dump_message_filter_ = NULL;
 654 }
 655
 656 void PeerConnectionDependencyFactory::EnsureWebRtcAudioDeviceImpl() {
 657   if (audio_device_.get())
 658     return;
 659
 660   audio_device_ = new WebRtcAudioDeviceImpl();
 661 }
 662
 663 }  // namespace content