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/location.h"
  11 #include "base/strings/utf_string_conversions.h"
  12 #include "base/synchronization/waitable_event.h"
  13 #include "content/common/media/media_stream_messages.h"
  14 #include "content/public/common/content_switches.h"
  15 #include "content/public/common/renderer_preferences.h"
  16 #include "content/renderer/media/media_stream.h"
  17 #include "content/renderer/media/media_stream_audio_processor.h"
  18 #include "content/renderer/media/media_stream_audio_processor_options.h"
  19 #include "content/renderer/media/media_stream_audio_source.h"
  20 #include "content/renderer/media/media_stream_video_source.h"
  21 #include "content/renderer/media/media_stream_video_track.h"
  22 #include "content/renderer/media/peer_connection_identity_store.h"
  23 #include "content/renderer/media/rtc_media_constraints.h"
  24 #include "content/renderer/media/rtc_peer_connection_handler.h"
  25 #include "content/renderer/media/rtc_video_decoder_factory.h"
  26 #include "content/renderer/media/rtc_video_encoder_factory.h"
  27 #include "content/renderer/media/webaudio_capturer_source.h"
  28 #include "content/renderer/media/webrtc/stun_field_trial.h"
  29 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
  30 #include "content/renderer/media/webrtc/webrtc_video_capturer_adapter.h"
  31 #include "content/renderer/media/webrtc_audio_device_impl.h"
  32 #include "content/renderer/media/webrtc_local_audio_track.h"
  33 #include "content/renderer/media/webrtc_logging.h"
  34 #include "content/renderer/media/webrtc_uma_histograms.h"
  35 #include "content/renderer/p2p/ipc_network_manager.h"
  36 #include "content/renderer/p2p/ipc_socket_factory.h"
  37 #include "content/renderer/p2p/port_allocator.h"
  38 #include "content/renderer/render_thread_impl.h"
  39 #include "content/renderer/render_view_impl.h"
  40 #include "jingle/glue/thread_wrapper.h"
  41 #include "media/renderers/gpu_video_accelerator_factories.h"
  42 #include "third_party/WebKit/public/platform/WebMediaConstraints.h"
  43 #include "third_party/WebKit/public/platform/WebMediaStream.h"
  44 #include "third_party/WebKit/public/platform/WebMediaStreamSource.h"
  45 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h"
  46 #include "third_party/WebKit/public/platform/WebURL.h"
  47 #include "third_party/WebKit/public/web/WebDocument.h"
  48 #include "third_party/WebKit/public/web/WebFrame.h"
  49 #include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface.h"
  50
  51 #if defined(USE_OPENSSL)
  52 #include "third_party/webrtc/base/ssladapter.h"
  53 #else
  54 #include "net/socket/nss_ssl_util.h"
  55 #endif
  56
  57 #if defined(OS_ANDROID)
  58 #include "media/base/android/media_codec_bridge.h"
  59 #endif
  60
  61 namespace content {
  62
  63 // Map of corresponding media constraints and platform effects.
  64 struct {
  65   const char* constraint;
  66   const media::AudioParameters::PlatformEffectsMask effect;
  67 } const kConstraintEffectMap[] = {
  68   { webrtc::MediaConstraintsInterface::kGoogEchoCancellation,
  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       }
 105     }
 106   }
 107 }
 108
 109 class P2PPortAllocatorFactory : public webrtc::PortAllocatorFactoryInterface {
 110  public:
 111   P2PPortAllocatorFactory(P2PSocketDispatcher* socket_dispatcher,
 112                           rtc::NetworkManager* network_manager,
 113                           rtc::PacketSocketFactory* socket_factory,
 114                           const GURL& origin,
 115                           const P2PPortAllocator::Config& config)
 116       : socket_dispatcher_(socket_dispatcher),
 117         network_manager_(network_manager),
 118         socket_factory_(socket_factory),
 119         origin_(origin),
 120         config_(config) {}
 121
 122   cricket::PortAllocator* CreatePortAllocator(
 123       const std::vector<StunConfiguration>& stun_servers,
 124       const std::vector<TurnConfiguration>& turn_configurations) override {
 125     P2PPortAllocator::Config config = config_;
 126     for (size_t i = 0; i < stun_servers.size(); ++i) {
 127       config.stun_servers.insert(rtc::SocketAddress(
 128           stun_servers[i].server.hostname(),
 129           stun_servers[i].server.port()));
 130     }
 131     for (size_t i = 0; i < turn_configurations.size(); ++i) {
 132       P2PPortAllocator::Config::RelayServerConfig relay_config;
 133       relay_config.server_address = turn_configurations[i].server.hostname();
 134       relay_config.port = turn_configurations[i].server.port();
 135       relay_config.username = turn_configurations[i].username;
 136       relay_config.password = turn_configurations[i].password;
 137       relay_config.transport_type = turn_configurations[i].transport_type;
 138       relay_config.secure = turn_configurations[i].secure;
 139       config.relays.push_back(relay_config);
 140     }
 141
 142     return new P2PPortAllocator(
 143         socket_dispatcher_.get(), network_manager_,
 144         socket_factory_, config, origin_);
 145   }
 146
 147  protected:
 148   ~P2PPortAllocatorFactory() override {}
 149
 150  private:
 151   scoped_refptr<P2PSocketDispatcher> socket_dispatcher_;
 152   // |network_manager_| and |socket_factory_| are a weak references, owned by
 153   // PeerConnectionDependencyFactory.
 154   rtc::NetworkManager* network_manager_;
 155   rtc::PacketSocketFactory* socket_factory_;
 156   // The origin URL of the WebFrame that created the
 157   // P2PPortAllocatorFactory.
 158   GURL origin_;
 159
 160   // Keep track of configuration common to all PortAllocators created by this
 161   // factory; additional, per-allocator configuration is passed into
 162   // CreatePortAllocator.
 163   P2PPortAllocator::Config config_;
 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   TryScheduleStunProbeTrial();
 175 }
 176
 177 PeerConnectionDependencyFactory::~PeerConnectionDependencyFactory() {
 178   DVLOG(1) << "~PeerConnectionDependencyFactory()";
 179   DCHECK(pc_factory_ == NULL);
 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_frame_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(CreateAudioCapturer(
 211       render_frame_id, device_info, audio_constraints, 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_.task_runner()->PostTask(
 296       FROM_HERE,
 297       base::Bind(&PeerConnectionDependencyFactory::InitializeWorkerThread,
 298                  base::Unretained(this), &worker_thread_, &start_worker_event));
 299
 300   base::WaitableEvent create_network_manager_event(true, false);
 301   chrome_worker_thread_.task_runner()->PostTask(
 302       FROM_HERE,
 303       base::Bind(&PeerConnectionDependencyFactory::
 304                      CreateIpcNetworkManagerOnWorkerThread,
 305                  base::Unretained(this), &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_.task_runner()->PostTask(
 326       FROM_HERE,
 327       base::Bind(&PeerConnectionDependencyFactory::InitializeSignalingThread,
 328                  base::Unretained(this),
 329                  RenderThreadImpl::current()->GetGpuFactories(),
 330                  &start_signaling_event));
 331
 332   start_signaling_event.Wait();
 333   CHECK(signaling_thread_);
 334 }
 335
 336 void PeerConnectionDependencyFactory::InitializeSignalingThread(
 337     const scoped_refptr<media::GpuVideoAcceleratorFactories>& gpu_factories,
 338     base::WaitableEvent* event) {
 339   DCHECK(chrome_signaling_thread_.task_runner()->BelongsToCurrentThread());
 340   DCHECK(worker_thread_);
 341   DCHECK(p2p_socket_dispatcher_.get());
 342
 343   jingle_glue::JingleThreadWrapper::EnsureForCurrentMessageLoop();
 344   jingle_glue::JingleThreadWrapper::current()->set_send_allowed(true);
 345   signaling_thread_ = jingle_glue::JingleThreadWrapper::current();
 346
 347   EnsureWebRtcAudioDeviceImpl();
 348
 349   socket_factory_.reset(
 350       new IpcPacketSocketFactory(p2p_socket_dispatcher_.get()));
 351
 352   scoped_ptr<cricket::WebRtcVideoDecoderFactory> decoder_factory;
 353   scoped_ptr<cricket::WebRtcVideoEncoderFactory> encoder_factory;
 354
 355   const base::CommandLine* cmd_line = base::CommandLine::ForCurrentProcess();
 356   if (gpu_factories && gpu_factories->IsGpuVideoAcceleratorEnabled()) {
 357     if (!cmd_line->HasSwitch(switches::kDisableWebRtcHWDecoding))
 358       decoder_factory.reset(new RTCVideoDecoderFactory(gpu_factories));
 359
 360     if (!cmd_line->HasSwitch(switches::kDisableWebRtcHWEncoding))
 361       encoder_factory.reset(new RTCVideoEncoderFactory(gpu_factories));
 362   }
 363
 364 #if defined(OS_ANDROID)
 365   if (!media::MediaCodecBridge::SupportsSetParameters())
 366     encoder_factory.reset();
 367 #endif
 368
 369   pc_factory_ = webrtc::CreatePeerConnectionFactory(
 370       worker_thread_, signaling_thread_, audio_device_.get(),
 371       encoder_factory.release(), decoder_factory.release());
 372   CHECK(pc_factory_.get());
 373
 374   webrtc::PeerConnectionFactoryInterface::Options factory_options;
 375   factory_options.disable_sctp_data_channels = false;
 376   factory_options.disable_encryption =
 377       cmd_line->HasSwitch(switches::kDisableWebRtcEncryption);
 378   if (cmd_line->HasSwitch(switches::kEnableWebRtcDtls12))
 379     factory_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
 380   pc_factory_->SetOptions(factory_options);
 381
 382   event->Signal();
 383 }
 384
 385 bool PeerConnectionDependencyFactory::PeerConnectionFactoryCreated() {
 386   return pc_factory_.get() != NULL;
 387 }
 388
 389 scoped_refptr<webrtc::PeerConnectionInterface>
 390 PeerConnectionDependencyFactory::CreatePeerConnection(
 391     const webrtc::PeerConnectionInterface::RTCConfiguration& config,
 392     const webrtc::MediaConstraintsInterface* constraints,
 393     blink::WebFrame* web_frame,
 394     webrtc::PeerConnectionObserver* observer) {
 395   CHECK(web_frame);
 396   CHECK(observer);
 397   if (!GetPcFactory().get())
 398     return NULL;
 399
 400   rtc::scoped_ptr<PeerConnectionIdentityStore> identity_store(
 401       new PeerConnectionIdentityStore(
 402           GURL(web_frame->document().url()),
 403           GURL(web_frame->document().firstPartyForCookies())));
 404
 405   // Copy the flag from Preference associated with this WebFrame.
 406   P2PPortAllocator::Config pref_config;
 407   if (web_frame && web_frame->view()) {
 408     RenderViewImpl* renderer_view_impl =
 409         RenderViewImpl::FromWebView(web_frame->view());
 410     if (renderer_view_impl) {
 411       pref_config.enable_multiple_routes =
 412           renderer_view_impl->renderer_preferences()
 413               .enable_webrtc_multiple_routes;
 414       pref_config.enable_nonproxied_udp =
 415           renderer_view_impl->renderer_preferences()
 416               .enable_webrtc_nonproxied_udp;
 417     }
 418   }
 419
 420   scoped_refptr<P2PPortAllocatorFactory> pa_factory =
 421       new rtc::RefCountedObject<P2PPortAllocatorFactory>(
 422           p2p_socket_dispatcher_.get(), network_manager_, socket_factory_.get(),
 423           GURL(web_frame->document().url().spec()).GetOrigin(), pref_config);
 424
 425   return GetPcFactory()->CreatePeerConnection(config,
 426                                               constraints,
 427                                               pa_factory.get(),
 428                                               identity_store.Pass(),
 429                                               observer).get();
 430 }
 431
 432 scoped_refptr<webrtc::MediaStreamInterface>
 433 PeerConnectionDependencyFactory::CreateLocalMediaStream(
 434     const std::string& label) {
 435   return GetPcFactory()->CreateLocalMediaStream(label).get();
 436 }
 437
 438 scoped_refptr<webrtc::AudioSourceInterface>
 439 PeerConnectionDependencyFactory::CreateLocalAudioSource(
 440     const webrtc::MediaConstraintsInterface* constraints) {
 441   scoped_refptr<webrtc::AudioSourceInterface> source =
 442       GetPcFactory()->CreateAudioSource(constraints).get();
 443   return source;
 444 }
 445
 446 void PeerConnectionDependencyFactory::CreateLocalAudioTrack(
 447     const blink::WebMediaStreamTrack& track) {
 448   blink::WebMediaStreamSource source = track.source();
 449   DCHECK_EQ(source.type(), blink::WebMediaStreamSource::TypeAudio);
 450   MediaStreamAudioSource* source_data =
 451       static_cast<MediaStreamAudioSource*>(source.extraData());
 452
 453   scoped_refptr<WebAudioCapturerSource> webaudio_source;
 454   if (!source_data) {
 455     if (source.requiresAudioConsumer()) {
 456       // We're adding a WebAudio MediaStream.
 457       // Create a specific capturer for each WebAudio consumer.
 458       webaudio_source = CreateWebAudioSource(&source);
 459       source_data =
 460           static_cast<MediaStreamAudioSource*>(source.extraData());
 461     } else {
 462       // TODO(perkj): Implement support for sources from
 463       // remote MediaStreams.
 464       NOTIMPLEMENTED();
 465       return;
 466     }
 467   }
 468
 469   // Creates an adapter to hold all the libjingle objects.
 470   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
 471       WebRtcLocalAudioTrackAdapter::Create(track.id().utf8(),
 472                                            source_data->local_audio_source()));
 473   static_cast<webrtc::AudioTrackInterface*>(adapter.get())->set_enabled(
 474       track.isEnabled());
 475
 476   // TODO(xians): Merge |source| to the capturer(). We can't do this today
 477   // because only one capturer() is supported while one |source| is created
 478   // for each audio track.
 479   scoped_ptr<WebRtcLocalAudioTrack> audio_track(new WebRtcLocalAudioTrack(
 480       adapter.get(), source_data->GetAudioCapturer(), webaudio_source.get()));
 481
 482   StartLocalAudioTrack(audio_track.get());
 483
 484   // Pass the ownership of the native local audio track to the blink track.
 485   blink::WebMediaStreamTrack writable_track = track;
 486   writable_track.setExtraData(audio_track.release());
 487 }
 488
 489 void PeerConnectionDependencyFactory::StartLocalAudioTrack(
 490     WebRtcLocalAudioTrack* audio_track) {
 491   // Start the audio track. This will hook the |audio_track| to the capturer
 492   // as the sink of the audio, and only start the source of the capturer if
 493   // it is the first audio track connecting to the capturer.
 494   audio_track->Start();
 495 }
 496
 497 scoped_refptr<WebAudioCapturerSource>
 498 PeerConnectionDependencyFactory::CreateWebAudioSource(
 499     blink::WebMediaStreamSource* source) {
 500   DVLOG(1) << "PeerConnectionDependencyFactory::CreateWebAudioSource()";
 501
 502   scoped_refptr<WebAudioCapturerSource>
 503       webaudio_capturer_source(new WebAudioCapturerSource(*source));
 504   MediaStreamAudioSource* source_data = new MediaStreamAudioSource();
 505
 506   // Use the current default capturer for the WebAudio track so that the
 507   // WebAudio track can pass a valid delay value and |need_audio_processing|
 508   // flag to PeerConnection.
 509   // TODO(xians): Remove this after moving APM to Chrome.
 510   if (GetWebRtcAudioDevice()) {
 511     source_data->SetAudioCapturer(
 512         GetWebRtcAudioDevice()->GetDefaultCapturer());
 513   }
 514
 515   // Create a LocalAudioSource object which holds audio options.
 516   // SetLocalAudioSource() affects core audio parts in third_party/Libjingle.
 517   source_data->SetLocalAudioSource(CreateLocalAudioSource(NULL).get());
 518   source->setExtraData(source_data);
 519
 520   // Replace the default source with WebAudio as source instead.
 521   source->addAudioConsumer(webaudio_capturer_source.get());
 522
 523   return webaudio_capturer_source;
 524 }
 525
 526 scoped_refptr<webrtc::VideoTrackInterface>
 527 PeerConnectionDependencyFactory::CreateLocalVideoTrack(
 528     const std::string& id,
 529     webrtc::VideoSourceInterface* source) {
 530   return GetPcFactory()->CreateVideoTrack(id, source).get();
 531 }
 532
 533 scoped_refptr<webrtc::VideoTrackInterface>
 534 PeerConnectionDependencyFactory::CreateLocalVideoTrack(
 535     const std::string& id, cricket::VideoCapturer* capturer) {
 536   if (!capturer) {
 537     LOG(ERROR) << "CreateLocalVideoTrack called with null VideoCapturer.";
 538     return NULL;
 539   }
 540
 541   // Create video source from the |capturer|.
 542   scoped_refptr<webrtc::VideoSourceInterface> source =
 543       GetPcFactory()->CreateVideoSource(capturer, NULL).get();
 544
 545   // Create native track from the source.
 546   return GetPcFactory()->CreateVideoTrack(id, source.get()).get();
 547 }
 548
 549 webrtc::SessionDescriptionInterface*
 550 PeerConnectionDependencyFactory::CreateSessionDescription(
 551     const std::string& type,
 552     const std::string& sdp,
 553     webrtc::SdpParseError* error) {
 554   return webrtc::CreateSessionDescription(type, sdp, error);
 555 }
 556
 557 webrtc::IceCandidateInterface*
 558 PeerConnectionDependencyFactory::CreateIceCandidate(
 559     const std::string& sdp_mid,
 560     int sdp_mline_index,
 561     const std::string& sdp) {
 562   return webrtc::CreateIceCandidate(sdp_mid, sdp_mline_index, sdp, nullptr);
 563 }
 564
 565 WebRtcAudioDeviceImpl*
 566 PeerConnectionDependencyFactory::GetWebRtcAudioDevice() {
 567   return audio_device_.get();
 568 }
 569
 570 void PeerConnectionDependencyFactory::InitializeWorkerThread(
 571     rtc::Thread** thread,
 572     base::WaitableEvent* event) {
 573   jingle_glue::JingleThreadWrapper::EnsureForCurrentMessageLoop();
 574   jingle_glue::JingleThreadWrapper::current()->set_send_allowed(true);
 575   *thread = jingle_glue::JingleThreadWrapper::current();
 576   event->Signal();
 577 }
 578
 579 void PeerConnectionDependencyFactory::TryScheduleStunProbeTrial() {
 580   const base::CommandLine* cmd_line = base::CommandLine::ForCurrentProcess();
 581
 582   if (!cmd_line->HasSwitch(switches::kWebRtcStunProbeTrialParameter))
 583     return;
 584
 585   GetPcFactory();
 586
 587   // The underneath IPC channel has to be connected before sending any IPC
 588   // message.
 589   if (!p2p_socket_dispatcher_->connected()) {
 590     base::MessageLoop::current()->PostDelayedTask(
 591         FROM_HERE,
 592         base::Bind(&PeerConnectionDependencyFactory::TryScheduleStunProbeTrial,
 593                    base::Unretained(this)),
 594         base::TimeDelta::FromSeconds(1));
 595     return;
 596   }
 597
 598   const std::string params =
 599       cmd_line->GetSwitchValueASCII(switches::kWebRtcStunProbeTrialParameter);
 600
 601   chrome_worker_thread_.task_runner()->PostDelayedTask(
 602       FROM_HERE,
 603       base::Bind(
 604           &PeerConnectionDependencyFactory::StartStunProbeTrialOnWorkerThread,
 605           base::Unretained(this), params),
 606       base::TimeDelta::FromMilliseconds(kExperimentStartDelayMs));
 607 }
 608
 609 void PeerConnectionDependencyFactory::StartStunProbeTrialOnWorkerThread(
 610     const std::string& params) {
 611   DCHECK(chrome_worker_thread_.task_runner()->BelongsToCurrentThread());
 612   rtc::NetworkManager::NetworkList networks;
 613   network_manager_->GetNetworks(&networks);
 614   stun_prober_ = StartStunProbeTrial(networks, params, socket_factory_.get());
 615 }
 616
 617 void PeerConnectionDependencyFactory::CreateIpcNetworkManagerOnWorkerThread(
 618     base::WaitableEvent* event) {
 619   DCHECK(chrome_worker_thread_.task_runner()->BelongsToCurrentThread());
 620   network_manager_ = new IpcNetworkManager(p2p_socket_dispatcher_.get());
 621   event->Signal();
 622 }
 623
 624 void PeerConnectionDependencyFactory::DeleteIpcNetworkManager() {
 625   DCHECK(chrome_worker_thread_.task_runner()->BelongsToCurrentThread());
 626   delete network_manager_;
 627   network_manager_ = NULL;
 628 }
 629
 630 void PeerConnectionDependencyFactory::CleanupPeerConnectionFactory() {
 631   DVLOG(1) << "PeerConnectionDependencyFactory::CleanupPeerConnectionFactory()";
 632   pc_factory_ = NULL;
 633   if (network_manager_) {
 634     // The network manager needs to free its resources on the thread they were
 635     // created, which is the worked thread.
 636     if (chrome_worker_thread_.IsRunning()) {
 637       chrome_worker_thread_.task_runner()->PostTask(
 638           FROM_HERE,
 639           base::Bind(&PeerConnectionDependencyFactory::DeleteIpcNetworkManager,
 640                      base::Unretained(this)));
 641       // Stopping the thread will wait until all tasks have been
 642       // processed before returning. We wait for the above task to finish before
 643       // letting the the function continue to avoid any potential race issues.
 644       chrome_worker_thread_.Stop();
 645     } else {
 646       NOTREACHED() << "Worker thread not running.";
 647     }
 648   }
 649 }
 650
 651 scoped_refptr<WebRtcAudioCapturer>
 652 PeerConnectionDependencyFactory::CreateAudioCapturer(
 653     int render_frame_id,
 654     const StreamDeviceInfo& device_info,
 655     const blink::WebMediaConstraints& constraints,
 656     MediaStreamAudioSource* audio_source) {
 657   // TODO(xians): Handle the cases when gUM is called without a proper render
 658   // view, for example, by an extension.
 659   DCHECK_GE(render_frame_id, 0);
 660
 661   EnsureWebRtcAudioDeviceImpl();
 662   DCHECK(GetWebRtcAudioDevice());
 663   return WebRtcAudioCapturer::CreateCapturer(
 664       render_frame_id, device_info, constraints, GetWebRtcAudioDevice(),
 665       audio_source);
 666 }
 667
 668 scoped_refptr<base::SingleThreadTaskRunner>
 669 PeerConnectionDependencyFactory::GetWebRtcWorkerThread() const {
 670   DCHECK(CalledOnValidThread());
 671   return chrome_worker_thread_.IsRunning() ? chrome_worker_thread_.task_runner()
 672                                            : nullptr;
 673 }
 674
 675 scoped_refptr<base::SingleThreadTaskRunner>
 676 PeerConnectionDependencyFactory::GetWebRtcSignalingThread() const {
 677   DCHECK(CalledOnValidThread());
 678   return chrome_signaling_thread_.IsRunning()
 679              ? chrome_signaling_thread_.task_runner()
 680              : nullptr;
 681 }
 682
 683 void PeerConnectionDependencyFactory::EnsureWebRtcAudioDeviceImpl() {
 684   if (audio_device_.get())
 685     return;
 686
 687   audio_device_ = new WebRtcAudioDeviceImpl();
 688 }
 689
 690 }  // namespace content