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_service.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   { content::kMediaStreamAudioDucking,
  69     media::AudioParameters::DUCKING },
  70   { webrtc::MediaConstraintsInterface::kGoogEchoCancellation,
  71     media::AudioParameters::ECHO_CANCELLER },
  72 };
  73
  74 // If any platform effects are available, check them against the constraints.
  75 // Disable effects to match false constraints, but if a constraint is true, set
  76 // the constraint to false to later disable the software effect.
  77 //
  78 // This function may modify both |constraints| and |effects|.
  79 void HarmonizeConstraintsAndEffects(RTCMediaConstraints* constraints,
  80                                     int* effects) {
  81   if (*effects != media::AudioParameters::NO_EFFECTS) {
  82     for (size_t i = 0; i < arraysize(kConstraintEffectMap); ++i) {
  83       bool value;
  84       size_t is_mandatory = 0;
  85       if (!webrtc::FindConstraint(constraints,
  86                                   kConstraintEffectMap[i].constraint,
  87                                   &value,
  88                                   &is_mandatory) || !value) {
  89         // If the constraint is false, or does not exist, disable the platform
  90         // effect.
  91         *effects &= ~kConstraintEffectMap[i].effect;
  92         DVLOG(1) << "Disabling platform effect: "
  93                  << kConstraintEffectMap[i].effect;
  94       } else if (*effects & kConstraintEffectMap[i].effect) {
  95         // If the constraint is true, leave the platform effect enabled, and
  96         // set the constraint to false to later disable the software effect.
  97         if (is_mandatory) {
  98           constraints->AddMandatory(kConstraintEffectMap[i].constraint,
  99               webrtc::MediaConstraintsInterface::kValueFalse, true);
 100         } else {
 101           constraints->AddOptional(kConstraintEffectMap[i].constraint,
 102               webrtc::MediaConstraintsInterface::kValueFalse, true);
 103         }
 104         DVLOG(1) << "Disabling constraint: "
 105                  << kConstraintEffectMap[i].constraint;
 106       } else if (kConstraintEffectMap[i].effect ==
 107                  media::AudioParameters::DUCKING && value && !is_mandatory) {
 108         // Special handling of the DUCKING flag that sets the optional
 109         // constraint to |false| to match what the device will support.
 110         constraints->AddOptional(kConstraintEffectMap[i].constraint,
 111             webrtc::MediaConstraintsInterface::kValueFalse, true);
 112         // No need to modify |effects| since the ducking flag is already off.
 113         DCHECK((*effects & media::AudioParameters::DUCKING) == 0);
 114       }
 115     }
 116   }
 117 }
 118
 119 class P2PPortAllocatorFactory : public webrtc::PortAllocatorFactoryInterface {
 120  public:
 121   P2PPortAllocatorFactory(P2PSocketDispatcher* socket_dispatcher,
 122                           rtc::NetworkManager* network_manager,
 123                           rtc::PacketSocketFactory* socket_factory,
 124                           const GURL& origin,
 125                           bool enable_multiple_routes)
 126       : socket_dispatcher_(socket_dispatcher),
 127         network_manager_(network_manager),
 128         socket_factory_(socket_factory),
 129         origin_(origin),
 130         enable_multiple_routes_(enable_multiple_routes) {}
 131
 132   cricket::PortAllocator* CreatePortAllocator(
 133       const std::vector<StunConfiguration>& stun_servers,
 134       const std::vector<TurnConfiguration>& turn_configurations) override {
 135     P2PPortAllocator::Config config;
 136     for (size_t i = 0; i < stun_servers.size(); ++i) {
 137       config.stun_servers.insert(rtc::SocketAddress(
 138           stun_servers[i].server.hostname(),
 139           stun_servers[i].server.port()));
 140     }
 141     for (size_t i = 0; i < turn_configurations.size(); ++i) {
 142       P2PPortAllocator::Config::RelayServerConfig relay_config;
 143       relay_config.server_address = turn_configurations[i].server.hostname();
 144       relay_config.port = turn_configurations[i].server.port();
 145       relay_config.username = turn_configurations[i].username;
 146       relay_config.password = turn_configurations[i].password;
 147       relay_config.transport_type = turn_configurations[i].transport_type;
 148       relay_config.secure = turn_configurations[i].secure;
 149       config.relays.push_back(relay_config);
 150
 151       // Use turn servers as stun servers.
 152       config.stun_servers.insert(rtc::SocketAddress(
 153           turn_configurations[i].server.hostname(),
 154           turn_configurations[i].server.port()));
 155     }
 156     config.enable_multiple_routes = enable_multiple_routes_;
 157
 158     return new P2PPortAllocator(
 159         socket_dispatcher_.get(), network_manager_,
 160         socket_factory_, config, origin_);
 161   }
 162
 163  protected:
 164   ~P2PPortAllocatorFactory() override {}
 165
 166  private:
 167   scoped_refptr<P2PSocketDispatcher> socket_dispatcher_;
 168   // |network_manager_| and |socket_factory_| are a weak references, owned by
 169   // PeerConnectionDependencyFactory.
 170   rtc::NetworkManager* network_manager_;
 171   rtc::PacketSocketFactory* socket_factory_;
 172   // The origin URL of the WebFrame that created the
 173   // P2PPortAllocatorFactory.
 174   GURL origin_;
 175   // When false, only 'any' address (all 0s) will be bound for address
 176   // discovery.
 177   bool enable_multiple_routes_;
 178 };
 179
 180 PeerConnectionDependencyFactory::PeerConnectionDependencyFactory(
 181     P2PSocketDispatcher* p2p_socket_dispatcher)
 182     : network_manager_(NULL),
 183       p2p_socket_dispatcher_(p2p_socket_dispatcher),
 184       signaling_thread_(NULL),
 185       worker_thread_(NULL),
 186       chrome_signaling_thread_("Chrome_libJingle_Signaling"),
 187       chrome_worker_thread_("Chrome_libJingle_WorkerThread") {
 188   TryScheduleStunProbeTrial();
 189 }
 190
 191 PeerConnectionDependencyFactory::~PeerConnectionDependencyFactory() {
 192   DVLOG(1) << "~PeerConnectionDependencyFactory()";
 193   DCHECK(pc_factory_ == NULL);
 194 }
 195
 196 blink::WebRTCPeerConnectionHandler*
 197 PeerConnectionDependencyFactory::CreateRTCPeerConnectionHandler(
 198     blink::WebRTCPeerConnectionHandlerClient* client) {
 199   // Save histogram data so we can see how much PeerConnetion is used.
 200   // The histogram counts the number of calls to the JS API
 201   // webKitRTCPeerConnection.
 202   UpdateWebRTCMethodCount(WEBKIT_RTC_PEER_CONNECTION);
 203
 204   return new RTCPeerConnectionHandler(client, this);
 205 }
 206
 207 bool PeerConnectionDependencyFactory::InitializeMediaStreamAudioSource(
 208     int render_frame_id,
 209     const blink::WebMediaConstraints& audio_constraints,
 210     MediaStreamAudioSource* source_data) {
 211   DVLOG(1) << "InitializeMediaStreamAudioSources()";
 212
 213   // Do additional source initialization if the audio source is a valid
 214   // microphone or tab audio.
 215   RTCMediaConstraints native_audio_constraints(audio_constraints);
 216   MediaAudioConstraints::ApplyFixedAudioConstraints(&native_audio_constraints);
 217
 218   StreamDeviceInfo device_info = source_data->device_info();
 219   RTCMediaConstraints constraints = native_audio_constraints;
 220   // May modify both |constraints| and |effects|.
 221   HarmonizeConstraintsAndEffects(&constraints,
 222                                  &device_info.device.input.effects);
 223
 224   scoped_refptr<WebRtcAudioCapturer> capturer(CreateAudioCapturer(
 225       render_frame_id, device_info, audio_constraints, source_data));
 226   if (!capturer.get()) {
 227     const std::string log_string =
 228         "PCDF::InitializeMediaStreamAudioSource: fails to create capturer";
 229     WebRtcLogMessage(log_string);
 230     DVLOG(1) << log_string;
 231     // TODO(xians): Don't we need to check if source_observer is observing
 232     // something? If not, then it looks like we have a leak here.
 233     // OTOH, if it _is_ observing something, then the callback might
 234     // be called multiple times which is likely also a bug.
 235     return false;
 236   }
 237   source_data->SetAudioCapturer(capturer.get());
 238
 239   // Creates a LocalAudioSource object which holds audio options.
 240   // TODO(xians): The option should apply to the track instead of the source.
 241   // TODO(perkj): Move audio constraints parsing to Chrome.
 242   // Currently there are a few constraints that are parsed by libjingle and
 243   // the state is set to ended if parsing fails.
 244   scoped_refptr<webrtc::AudioSourceInterface> rtc_source(
 245       CreateLocalAudioSource(&constraints).get());
 246   if (rtc_source->state() != webrtc::MediaSourceInterface::kLive) {
 247     DLOG(WARNING) << "Failed to create rtc LocalAudioSource.";
 248     return false;
 249   }
 250   source_data->SetLocalAudioSource(rtc_source.get());
 251   return true;
 252 }
 253
 254 WebRtcVideoCapturerAdapter*
 255 PeerConnectionDependencyFactory::CreateVideoCapturer(
 256     bool is_screeencast) {
 257   // We need to make sure the libjingle thread wrappers have been created
 258   // before we can use an instance of a WebRtcVideoCapturerAdapter. This is
 259   // since the base class of WebRtcVideoCapturerAdapter is a
 260   // cricket::VideoCapturer and it uses the libjingle thread wrappers.
 261   if (!GetPcFactory().get())
 262     return NULL;
 263   return new WebRtcVideoCapturerAdapter(is_screeencast);
 264 }
 265
 266 scoped_refptr<webrtc::VideoSourceInterface>
 267 PeerConnectionDependencyFactory::CreateVideoSource(
 268     cricket::VideoCapturer* capturer,
 269     const blink::WebMediaConstraints& constraints) {
 270   RTCMediaConstraints webrtc_constraints(constraints);
 271   scoped_refptr<webrtc::VideoSourceInterface> source =
 272       GetPcFactory()->CreateVideoSource(capturer, &webrtc_constraints).get();
 273   return source;
 274 }
 275
 276 const scoped_refptr<webrtc::PeerConnectionFactoryInterface>&
 277 PeerConnectionDependencyFactory::GetPcFactory() {
 278   if (!pc_factory_.get())
 279     CreatePeerConnectionFactory();
 280   CHECK(pc_factory_.get());
 281   return pc_factory_;
 282 }
 283
 284
 285 void PeerConnectionDependencyFactory::WillDestroyCurrentMessageLoop() {
 286   CleanupPeerConnectionFactory();
 287 }
 288
 289 void PeerConnectionDependencyFactory::CreatePeerConnectionFactory() {
 290   DCHECK(!pc_factory_.get());
 291   DCHECK(!signaling_thread_);
 292   DCHECK(!worker_thread_);
 293   DCHECK(!network_manager_);
 294   DCHECK(!socket_factory_);
 295   DCHECK(!chrome_signaling_thread_.IsRunning());
 296   DCHECK(!chrome_worker_thread_.IsRunning());
 297
 298   DVLOG(1) << "PeerConnectionDependencyFactory::CreatePeerConnectionFactory()";
 299
 300   base::MessageLoop::current()->AddDestructionObserver(this);
 301   // To allow sending to the signaling/worker threads.
 302   jingle_glue::JingleThreadWrapper::EnsureForCurrentMessageLoop();
 303   jingle_glue::JingleThreadWrapper::current()->set_send_allowed(true);
 304
 305   CHECK(chrome_signaling_thread_.Start());
 306   CHECK(chrome_worker_thread_.Start());
 307
 308   base::WaitableEvent start_worker_event(true, false);
 309   chrome_worker_thread_.task_runner()->PostTask(
 310       FROM_HERE,
 311       base::Bind(&PeerConnectionDependencyFactory::InitializeWorkerThread,
 312                  base::Unretained(this), &worker_thread_, &start_worker_event));
 313
 314   base::WaitableEvent create_network_manager_event(true, false);
 315   chrome_worker_thread_.task_runner()->PostTask(
 316       FROM_HERE,
 317       base::Bind(&PeerConnectionDependencyFactory::
 318                      CreateIpcNetworkManagerOnWorkerThread,
 319                  base::Unretained(this), &create_network_manager_event));
 320
 321   start_worker_event.Wait();
 322   create_network_manager_event.Wait();
 323
 324   CHECK(worker_thread_);
 325
 326   // Init SSL, which will be needed by PeerConnection.
 327 #if defined(USE_OPENSSL)
 328   if (!rtc::InitializeSSL()) {
 329     LOG(ERROR) << "Failed on InitializeSSL.";
 330     NOTREACHED();
 331     return;
 332   }
 333 #else
 334   // TODO(ronghuawu): Replace this call with InitializeSSL.
 335   net::EnsureNSSSSLInit();
 336 #endif
 337
 338   base::WaitableEvent start_signaling_event(true, false);
 339   chrome_signaling_thread_.task_runner()->PostTask(
 340       FROM_HERE,
 341       base::Bind(&PeerConnectionDependencyFactory::InitializeSignalingThread,
 342                  base::Unretained(this),
 343                  RenderThreadImpl::current()->GetGpuFactories(),
 344                  &start_signaling_event));
 345
 346   start_signaling_event.Wait();
 347   CHECK(signaling_thread_);
 348 }
 349
 350 void PeerConnectionDependencyFactory::InitializeSignalingThread(
 351     const scoped_refptr<media::GpuVideoAcceleratorFactories>& gpu_factories,
 352     base::WaitableEvent* event) {
 353   DCHECK(chrome_signaling_thread_.task_runner()->BelongsToCurrentThread());
 354   DCHECK(worker_thread_);
 355   DCHECK(p2p_socket_dispatcher_.get());
 356
 357   jingle_glue::JingleThreadWrapper::EnsureForCurrentMessageLoop();
 358   jingle_glue::JingleThreadWrapper::current()->set_send_allowed(true);
 359   signaling_thread_ = jingle_glue::JingleThreadWrapper::current();
 360
 361   EnsureWebRtcAudioDeviceImpl();
 362
 363   socket_factory_.reset(
 364       new IpcPacketSocketFactory(p2p_socket_dispatcher_.get()));
 365
 366   scoped_ptr<cricket::WebRtcVideoDecoderFactory> decoder_factory;
 367   scoped_ptr<cricket::WebRtcVideoEncoderFactory> encoder_factory;
 368
 369   const base::CommandLine* cmd_line = base::CommandLine::ForCurrentProcess();
 370   if (gpu_factories && gpu_factories->IsGpuVideoAcceleratorEnabled()) {
 371     if (!cmd_line->HasSwitch(switches::kDisableWebRtcHWDecoding))
 372       decoder_factory.reset(new RTCVideoDecoderFactory(gpu_factories));
 373
 374     if (!cmd_line->HasSwitch(switches::kDisableWebRtcHWEncoding))
 375       encoder_factory.reset(new RTCVideoEncoderFactory(gpu_factories));
 376   }
 377
 378 #if defined(OS_ANDROID)
 379   if (!media::MediaCodecBridge::SupportsSetParameters())
 380     encoder_factory.reset();
 381 #endif
 382
 383   pc_factory_ = webrtc::CreatePeerConnectionFactory(
 384       worker_thread_, signaling_thread_, audio_device_.get(),
 385       encoder_factory.release(), decoder_factory.release());
 386   CHECK(pc_factory_.get());
 387
 388   webrtc::PeerConnectionFactoryInterface::Options factory_options;
 389   factory_options.disable_sctp_data_channels = false;
 390   factory_options.disable_encryption =
 391       cmd_line->HasSwitch(switches::kDisableWebRtcEncryption);
 392   if (cmd_line->HasSwitch(switches::kEnableWebRtcDtls12))
 393     factory_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
 394   pc_factory_->SetOptions(factory_options);
 395
 396   event->Signal();
 397 }
 398
 399 bool PeerConnectionDependencyFactory::PeerConnectionFactoryCreated() {
 400   return pc_factory_.get() != NULL;
 401 }
 402
 403 scoped_refptr<webrtc::PeerConnectionInterface>
 404 PeerConnectionDependencyFactory::CreatePeerConnection(
 405     const webrtc::PeerConnectionInterface::RTCConfiguration& config,
 406     const webrtc::MediaConstraintsInterface* constraints,
 407     blink::WebFrame* web_frame,
 408     webrtc::PeerConnectionObserver* observer) {
 409   CHECK(web_frame);
 410   CHECK(observer);
 411   if (!GetPcFactory().get())
 412     return NULL;
 413
 414   // Copy the flag from Preference associated with this WebFrame.
 415   bool enable_multiple_routes = true;
 416   PeerConnectionIdentityService* identity_service =
 417       new PeerConnectionIdentityService(
 418           GURL(web_frame->document().url()),
 419           GURL(web_frame->document().firstPartyForCookies()));
 420
 421   if (web_frame && web_frame->view()) {
 422     RenderViewImpl* renderer_view_impl =
 423         RenderViewImpl::FromWebView(web_frame->view());
 424     if (renderer_view_impl) {
 425       enable_multiple_routes = renderer_view_impl->renderer_preferences()
 426                                     .enable_webrtc_multiple_routes;
 427     }
 428   }
 429
 430   scoped_refptr<P2PPortAllocatorFactory> pa_factory =
 431       new rtc::RefCountedObject<P2PPortAllocatorFactory>(
 432           p2p_socket_dispatcher_.get(), network_manager_, socket_factory_.get(),
 433           GURL(web_frame->document().url().spec()).GetOrigin(),
 434           enable_multiple_routes);
 435
 436   return GetPcFactory()->CreatePeerConnection(config,
 437                                               constraints,
 438                                               pa_factory.get(),
 439                                               identity_service,
 440                                               observer).get();
 441 }
 442
 443 scoped_refptr<webrtc::MediaStreamInterface>
 444 PeerConnectionDependencyFactory::CreateLocalMediaStream(
 445     const std::string& label) {
 446   return GetPcFactory()->CreateLocalMediaStream(label).get();
 447 }
 448
 449 scoped_refptr<webrtc::AudioSourceInterface>
 450 PeerConnectionDependencyFactory::CreateLocalAudioSource(
 451     const webrtc::MediaConstraintsInterface* constraints) {
 452   scoped_refptr<webrtc::AudioSourceInterface> source =
 453       GetPcFactory()->CreateAudioSource(constraints).get();
 454   return source;
 455 }
 456
 457 void PeerConnectionDependencyFactory::CreateLocalAudioTrack(
 458     const blink::WebMediaStreamTrack& track) {
 459   blink::WebMediaStreamSource source = track.source();
 460   DCHECK_EQ(source.type(), blink::WebMediaStreamSource::TypeAudio);
 461   MediaStreamAudioSource* source_data =
 462       static_cast<MediaStreamAudioSource*>(source.extraData());
 463
 464   scoped_refptr<WebAudioCapturerSource> webaudio_source;
 465   if (!source_data) {
 466     if (source.requiresAudioConsumer()) {
 467       // We're adding a WebAudio MediaStream.
 468       // Create a specific capturer for each WebAudio consumer.
 469       webaudio_source = CreateWebAudioSource(&source);
 470       source_data =
 471           static_cast<MediaStreamAudioSource*>(source.extraData());
 472     } else {
 473       // TODO(perkj): Implement support for sources from
 474       // remote MediaStreams.
 475       NOTIMPLEMENTED();
 476       return;
 477     }
 478   }
 479
 480   // Creates an adapter to hold all the libjingle objects.
 481   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
 482       WebRtcLocalAudioTrackAdapter::Create(track.id().utf8(),
 483                                            source_data->local_audio_source()));
 484   static_cast<webrtc::AudioTrackInterface*>(adapter.get())->set_enabled(
 485       track.isEnabled());
 486
 487   // TODO(xians): Merge |source| to the capturer(). We can't do this today
 488   // because only one capturer() is supported while one |source| is created
 489   // for each audio track.
 490   scoped_ptr<WebRtcLocalAudioTrack> audio_track(new WebRtcLocalAudioTrack(
 491       adapter.get(), source_data->GetAudioCapturer(), webaudio_source.get()));
 492
 493   StartLocalAudioTrack(audio_track.get());
 494
 495   // Pass the ownership of the native local audio track to the blink track.
 496   blink::WebMediaStreamTrack writable_track = track;
 497   writable_track.setExtraData(audio_track.release());
 498 }
 499
 500 void PeerConnectionDependencyFactory::StartLocalAudioTrack(
 501     WebRtcLocalAudioTrack* audio_track) {
 502   // Start the audio track. This will hook the |audio_track| to the capturer
 503   // as the sink of the audio, and only start the source of the capturer if
 504   // it is the first audio track connecting to the capturer.
 505   audio_track->Start();
 506 }
 507
 508 scoped_refptr<WebAudioCapturerSource>
 509 PeerConnectionDependencyFactory::CreateWebAudioSource(
 510     blink::WebMediaStreamSource* source) {
 511   DVLOG(1) << "PeerConnectionDependencyFactory::CreateWebAudioSource()";
 512
 513   scoped_refptr<WebAudioCapturerSource>
 514       webaudio_capturer_source(new WebAudioCapturerSource(*source));
 515   MediaStreamAudioSource* source_data = new MediaStreamAudioSource();
 516
 517   // Use the current default capturer for the WebAudio track so that the
 518   // WebAudio track can pass a valid delay value and |need_audio_processing|
 519   // flag to PeerConnection.
 520   // TODO(xians): Remove this after moving APM to Chrome.
 521   if (GetWebRtcAudioDevice()) {
 522     source_data->SetAudioCapturer(
 523         GetWebRtcAudioDevice()->GetDefaultCapturer());
 524   }
 525
 526   // Create a LocalAudioSource object which holds audio options.
 527   // SetLocalAudioSource() affects core audio parts in third_party/Libjingle.
 528   source_data->SetLocalAudioSource(CreateLocalAudioSource(NULL).get());
 529   source->setExtraData(source_data);
 530
 531   // Replace the default source with WebAudio as source instead.
 532   source->addAudioConsumer(webaudio_capturer_source.get());
 533
 534   return webaudio_capturer_source;
 535 }
 536
 537 scoped_refptr<webrtc::VideoTrackInterface>
 538 PeerConnectionDependencyFactory::CreateLocalVideoTrack(
 539     const std::string& id,
 540     webrtc::VideoSourceInterface* source) {
 541   return GetPcFactory()->CreateVideoTrack(id, source).get();
 542 }
 543
 544 scoped_refptr<webrtc::VideoTrackInterface>
 545 PeerConnectionDependencyFactory::CreateLocalVideoTrack(
 546     const std::string& id, cricket::VideoCapturer* capturer) {
 547   if (!capturer) {
 548     LOG(ERROR) << "CreateLocalVideoTrack called with null VideoCapturer.";
 549     return NULL;
 550   }
 551
 552   // Create video source from the |capturer|.
 553   scoped_refptr<webrtc::VideoSourceInterface> source =
 554       GetPcFactory()->CreateVideoSource(capturer, NULL).get();
 555
 556   // Create native track from the source.
 557   return GetPcFactory()->CreateVideoTrack(id, source.get()).get();
 558 }
 559
 560 webrtc::SessionDescriptionInterface*
 561 PeerConnectionDependencyFactory::CreateSessionDescription(
 562     const std::string& type,
 563     const std::string& sdp,
 564     webrtc::SdpParseError* error) {
 565   return webrtc::CreateSessionDescription(type, sdp, error);
 566 }
 567
 568 webrtc::IceCandidateInterface*
 569 PeerConnectionDependencyFactory::CreateIceCandidate(
 570     const std::string& sdp_mid,
 571     int sdp_mline_index,
 572     const std::string& sdp) {
 573   return webrtc::CreateIceCandidate(sdp_mid, sdp_mline_index, sdp, nullptr);
 574 }
 575
 576 WebRtcAudioDeviceImpl*
 577 PeerConnectionDependencyFactory::GetWebRtcAudioDevice() {
 578   return audio_device_.get();
 579 }
 580
 581 void PeerConnectionDependencyFactory::InitializeWorkerThread(
 582     rtc::Thread** thread,
 583     base::WaitableEvent* event) {
 584   jingle_glue::JingleThreadWrapper::EnsureForCurrentMessageLoop();
 585   jingle_glue::JingleThreadWrapper::current()->set_send_allowed(true);
 586   *thread = jingle_glue::JingleThreadWrapper::current();
 587   event->Signal();
 588 }
 589
 590 void PeerConnectionDependencyFactory::TryScheduleStunProbeTrial() {
 591   const base::CommandLine* cmd_line = base::CommandLine::ForCurrentProcess();
 592
 593   if (!cmd_line->HasSwitch(switches::kWebRtcStunProbeTrialParameter))
 594     return;
 595
 596   GetPcFactory();
 597
 598   // The underneath IPC channel has to be connected before sending any IPC
 599   // message.
 600   if (!p2p_socket_dispatcher_->connected()) {
 601     base::MessageLoop::current()->PostDelayedTask(
 602         FROM_HERE,
 603         base::Bind(&PeerConnectionDependencyFactory::TryScheduleStunProbeTrial,
 604                    base::Unretained(this)),
 605         base::TimeDelta::FromSeconds(1));
 606     return;
 607   }
 608
 609   const std::string params =
 610       cmd_line->GetSwitchValueASCII(switches::kWebRtcStunProbeTrialParameter);
 611
 612   chrome_worker_thread_.task_runner()->PostDelayedTask(
 613       FROM_HERE,
 614       base::Bind(
 615           &PeerConnectionDependencyFactory::StartStunProbeTrialOnWorkerThread,
 616           base::Unretained(this), params),
 617       base::TimeDelta::FromMilliseconds(kExperimentStartDelayMs));
 618 }
 619
 620 void PeerConnectionDependencyFactory::StartStunProbeTrialOnWorkerThread(
 621     const std::string& params) {
 622   DCHECK(chrome_worker_thread_.task_runner()->BelongsToCurrentThread());
 623   rtc::NetworkManager::NetworkList networks;
 624   network_manager_->GetNetworks(&networks);
 625   stun_prober_ = StartStunProbeTrial(networks, params, socket_factory_.get());
 626 }
 627
 628 void PeerConnectionDependencyFactory::CreateIpcNetworkManagerOnWorkerThread(
 629     base::WaitableEvent* event) {
 630   DCHECK(chrome_worker_thread_.task_runner()->BelongsToCurrentThread());
 631   network_manager_ = new IpcNetworkManager(p2p_socket_dispatcher_.get());
 632   event->Signal();
 633 }
 634
 635 void PeerConnectionDependencyFactory::DeleteIpcNetworkManager() {
 636   DCHECK(chrome_worker_thread_.task_runner()->BelongsToCurrentThread());
 637   delete network_manager_;
 638   network_manager_ = NULL;
 639 }
 640
 641 void PeerConnectionDependencyFactory::CleanupPeerConnectionFactory() {
 642   DVLOG(1) << "PeerConnectionDependencyFactory::CleanupPeerConnectionFactory()";
 643   pc_factory_ = NULL;
 644   if (network_manager_) {
 645     // The network manager needs to free its resources on the thread they were
 646     // created, which is the worked thread.
 647     if (chrome_worker_thread_.IsRunning()) {
 648       chrome_worker_thread_.task_runner()->PostTask(
 649           FROM_HERE,
 650           base::Bind(&PeerConnectionDependencyFactory::DeleteIpcNetworkManager,
 651                      base::Unretained(this)));
 652       // Stopping the thread will wait until all tasks have been
 653       // processed before returning. We wait for the above task to finish before
 654       // letting the the function continue to avoid any potential race issues.
 655       chrome_worker_thread_.Stop();
 656     } else {
 657       NOTREACHED() << "Worker thread not running.";
 658     }
 659   }
 660 }
 661
 662 scoped_refptr<WebRtcAudioCapturer>
 663 PeerConnectionDependencyFactory::CreateAudioCapturer(
 664     int render_frame_id,
 665     const StreamDeviceInfo& device_info,
 666     const blink::WebMediaConstraints& constraints,
 667     MediaStreamAudioSource* audio_source) {
 668   // TODO(xians): Handle the cases when gUM is called without a proper render
 669   // view, for example, by an extension.
 670   DCHECK_GE(render_frame_id, 0);
 671
 672   EnsureWebRtcAudioDeviceImpl();
 673   DCHECK(GetWebRtcAudioDevice());
 674   return WebRtcAudioCapturer::CreateCapturer(
 675       render_frame_id, device_info, constraints, GetWebRtcAudioDevice(),
 676       audio_source);
 677 }
 678
 679 scoped_refptr<base::SingleThreadTaskRunner>
 680 PeerConnectionDependencyFactory::GetWebRtcWorkerThread() const {
 681   DCHECK(CalledOnValidThread());
 682   return chrome_worker_thread_.IsRunning() ? chrome_worker_thread_.task_runner()
 683                                            : nullptr;
 684 }
 685
 686 scoped_refptr<base::SingleThreadTaskRunner>
 687 PeerConnectionDependencyFactory::GetWebRtcSignalingThread() const {
 688   DCHECK(CalledOnValidThread());
 689   return chrome_signaling_thread_.IsRunning()
 690              ? chrome_signaling_thread_.task_runner()
 691              : nullptr;
 692 }
 693
 694 void PeerConnectionDependencyFactory::EnsureWebRtcAudioDeviceImpl() {
 695   if (audio_device_.get())
 696     return;
 697
 698   audio_device_ = new WebRtcAudioDeviceImpl();
 699 }
 700
 701 }  // namespace content