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 57namespace content { 58 59// Map of corresponding media constraints and platform effects. 60struct { 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|. 75void HarmonizeConstraintsAndEffects(RTCMediaConstraints* constraints, 76 int* effects) { 77 if (*effects != media::AudioParameters::NO_EFFECTS) { 78 for (size_t i = 0; i < ARRAYSIZE_UNSAFE(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 115class P2PPortAllocatorFactory : public webrtc::PortAllocatorFactoryInterface { 116 public: 117 P2PPortAllocatorFactory( 118 P2PSocketDispatcher* socket_dispatcher, 119 rtc::NetworkManager* network_manager, 120 rtc::PacketSocketFactory* socket_factory, 121 blink::WebFrame* web_frame) 122 : socket_dispatcher_(socket_dispatcher), 123 network_manager_(network_manager), 124 socket_factory_(socket_factory), 125 web_frame_(web_frame) { 126 } 127 128 virtual cricket::PortAllocator* CreatePortAllocator( 129 const std::vector<StunConfiguration>& stun_servers, 130 const std::vector<TurnConfiguration>& turn_configurations) OVERRIDE { 131 CHECK(web_frame_); 132 P2PPortAllocator::Config config; 133 for (size_t i = 0; i < stun_servers.size(); ++i) { 134 config.stun_servers.insert(rtc::SocketAddress( 135 stun_servers[i].server.hostname(), 136 stun_servers[i].server.port())); 137 } 138 config.legacy_relay = false; 139 for (size_t i = 0; i < turn_configurations.size(); ++i) { 140 P2PPortAllocator::Config::RelayServerConfig relay_config; 141 relay_config.server_address = turn_configurations[i].server.hostname(); 142 relay_config.port = turn_configurations[i].server.port(); 143 relay_config.username = turn_configurations[i].username; 144 relay_config.password = turn_configurations[i].password; 145 relay_config.transport_type = turn_configurations[i].transport_type; 146 relay_config.secure = turn_configurations[i].secure; 147 config.relays.push_back(relay_config); 148 149 // Use turn servers as stun servers. 150 config.stun_servers.insert(rtc::SocketAddress( 151 turn_configurations[i].server.hostname(), 152 turn_configurations[i].server.port())); 153 } 154 155 return new P2PPortAllocator( 156 web_frame_, socket_dispatcher_.get(), network_manager_, 157 socket_factory_, config); 158 } 159 160 protected: 161 virtual ~P2PPortAllocatorFactory() {} 162 163 private: 164 scoped_refptr<P2PSocketDispatcher> socket_dispatcher_; 165 // |network_manager_| and |socket_factory_| are a weak references, owned by 166 // PeerConnectionDependencyFactory. 167 rtc::NetworkManager* network_manager_; 168 rtc::PacketSocketFactory* socket_factory_; 169 // Raw ptr to the WebFrame that created the P2PPortAllocatorFactory. 170 blink::WebFrame* web_frame_; 171}; 172 173PeerConnectionDependencyFactory::PeerConnectionDependencyFactory( 174 P2PSocketDispatcher* p2p_socket_dispatcher) 175 : network_manager_(NULL), 176 p2p_socket_dispatcher_(p2p_socket_dispatcher), 177 signaling_thread_(NULL), 178 worker_thread_(NULL), 179 chrome_worker_thread_("Chrome_libJingle_WorkerThread") { 180} 181 182PeerConnectionDependencyFactory::~PeerConnectionDependencyFactory() { 183 CleanupPeerConnectionFactory(); 184 if (aec_dump_message_filter_.get()) 185 aec_dump_message_filter_->RemoveDelegate(this); 186} 187 188blink::WebRTCPeerConnectionHandler* 189PeerConnectionDependencyFactory::CreateRTCPeerConnectionHandler( 190 blink::WebRTCPeerConnectionHandlerClient* client) { 191 // Save histogram data so we can see how much PeerConnetion is used. 192 // The histogram counts the number of calls to the JS API 193 // webKitRTCPeerConnection. 194 UpdateWebRTCMethodCount(WEBKIT_RTC_PEER_CONNECTION); 195 196 return new RTCPeerConnectionHandler(client, this); 197} 198 199bool PeerConnectionDependencyFactory::InitializeMediaStreamAudioSource( 200 int render_view_id, 201 const blink::WebMediaConstraints& audio_constraints, 202 MediaStreamAudioSource* source_data) { 203 DVLOG(1) << "InitializeMediaStreamAudioSources()"; 204 205 // Do additional source initialization if the audio source is a valid 206 // microphone or tab audio. 207 RTCMediaConstraints native_audio_constraints(audio_constraints); 208 MediaAudioConstraints::ApplyFixedAudioConstraints(&native_audio_constraints); 209 210 StreamDeviceInfo device_info = source_data->device_info(); 211 RTCMediaConstraints constraints = native_audio_constraints; 212 // May modify both |constraints| and |effects|. 213 HarmonizeConstraintsAndEffects(&constraints, 214 &device_info.device.input.effects); 215 216 scoped_refptr<WebRtcAudioCapturer> capturer( 217 CreateAudioCapturer(render_view_id, device_info, audio_constraints, 218 source_data)); 219 if (!capturer.get()) { 220 const std::string log_string = 221 "PCDF::InitializeMediaStreamAudioSource: fails to create capturer"; 222 WebRtcLogMessage(log_string); 223 DVLOG(1) << log_string; 224 // TODO(xians): Don't we need to check if source_observer is observing 225 // something? If not, then it looks like we have a leak here. 226 // OTOH, if it _is_ observing something, then the callback might 227 // be called multiple times which is likely also a bug. 228 return false; 229 } 230 source_data->SetAudioCapturer(capturer.get()); 231 232 // Creates a LocalAudioSource object which holds audio options. 233 // TODO(xians): The option should apply to the track instead of the source. 234 // TODO(perkj): Move audio constraints parsing to Chrome. 235 // Currently there are a few constraints that are parsed by libjingle and 236 // the state is set to ended if parsing fails. 237 scoped_refptr<webrtc::AudioSourceInterface> rtc_source( 238 CreateLocalAudioSource(&constraints).get()); 239 if (rtc_source->state() != webrtc::MediaSourceInterface::kLive) { 240 DLOG(WARNING) << "Failed to create rtc LocalAudioSource."; 241 return false; 242 } 243 source_data->SetLocalAudioSource(rtc_source.get()); 244 return true; 245} 246 247WebRtcVideoCapturerAdapter* 248PeerConnectionDependencyFactory::CreateVideoCapturer( 249 bool is_screeencast) { 250 // We need to make sure the libjingle thread wrappers have been created 251 // before we can use an instance of a WebRtcVideoCapturerAdapter. This is 252 // since the base class of WebRtcVideoCapturerAdapter is a 253 // cricket::VideoCapturer and it uses the libjingle thread wrappers. 254 if (!GetPcFactory().get()) 255 return NULL; 256 return new WebRtcVideoCapturerAdapter(is_screeencast); 257} 258 259scoped_refptr<webrtc::VideoSourceInterface> 260PeerConnectionDependencyFactory::CreateVideoSource( 261 cricket::VideoCapturer* capturer, 262 const blink::WebMediaConstraints& constraints) { 263 RTCMediaConstraints webrtc_constraints(constraints); 264 scoped_refptr<webrtc::VideoSourceInterface> source = 265 GetPcFactory()->CreateVideoSource(capturer, &webrtc_constraints).get(); 266 return source; 267} 268 269const scoped_refptr<webrtc::PeerConnectionFactoryInterface>& 270PeerConnectionDependencyFactory::GetPcFactory() { 271 if (!pc_factory_.get()) 272 CreatePeerConnectionFactory(); 273 CHECK(pc_factory_.get()); 274 return pc_factory_; 275} 276 277void PeerConnectionDependencyFactory::CreatePeerConnectionFactory() { 278 DCHECK(!pc_factory_.get()); 279 DCHECK(!signaling_thread_); 280 DCHECK(!worker_thread_); 281 DCHECK(!network_manager_); 282 DCHECK(!socket_factory_); 283 DCHECK(!chrome_worker_thread_.IsRunning()); 284 285 DVLOG(1) << "PeerConnectionDependencyFactory::CreatePeerConnectionFactory()"; 286 287 jingle_glue::JingleThreadWrapper::EnsureForCurrentMessageLoop(); 288 jingle_glue::JingleThreadWrapper::current()->set_send_allowed(true); 289 signaling_thread_ = jingle_glue::JingleThreadWrapper::current(); 290 CHECK(signaling_thread_); 291 292 CHECK(chrome_worker_thread_.Start()); 293 294 base::WaitableEvent start_worker_event(true, false); 295 chrome_worker_thread_.message_loop()->PostTask(FROM_HERE, base::Bind( 296 &PeerConnectionDependencyFactory::InitializeWorkerThread, 297 base::Unretained(this), 298 &worker_thread_, 299 &start_worker_event)); 300 start_worker_event.Wait(); 301 CHECK(worker_thread_); 302 303 base::WaitableEvent create_network_manager_event(true, false); 304 chrome_worker_thread_.message_loop()->PostTask(FROM_HERE, base::Bind( 305 &PeerConnectionDependencyFactory::CreateIpcNetworkManagerOnWorkerThread, 306 base::Unretained(this), 307 &create_network_manager_event)); 308 create_network_manager_event.Wait(); 309 310 socket_factory_.reset( 311 new IpcPacketSocketFactory(p2p_socket_dispatcher_.get())); 312 313 // Init SSL, which will be needed by PeerConnection. 314#if defined(USE_OPENSSL) 315 if (!rtc::InitializeSSL()) { 316 LOG(ERROR) << "Failed on InitializeSSL."; 317 NOTREACHED(); 318 return; 319 } 320#else 321 // TODO(ronghuawu): Replace this call with InitializeSSL. 322 net::EnsureNSSSSLInit(); 323#endif 324 325 scoped_ptr<cricket::WebRtcVideoDecoderFactory> decoder_factory; 326 scoped_ptr<cricket::WebRtcVideoEncoderFactory> encoder_factory; 327 328 const CommandLine* cmd_line = CommandLine::ForCurrentProcess(); 329 scoped_refptr<media::GpuVideoAcceleratorFactories> gpu_factories = 330 RenderThreadImpl::current()->GetGpuFactories(); 331 if (!cmd_line->HasSwitch(switches::kDisableWebRtcHWDecoding)) { 332 if (gpu_factories.get()) 333 decoder_factory.reset(new RTCVideoDecoderFactory(gpu_factories)); 334 } 335 336 if (!cmd_line->HasSwitch(switches::kDisableWebRtcHWEncoding)) { 337 if (gpu_factories.get()) 338 encoder_factory.reset(new RTCVideoEncoderFactory(gpu_factories)); 339 } 340 341#if defined(OS_ANDROID) 342 if (!media::MediaCodecBridge::SupportsSetParameters()) 343 encoder_factory.reset(); 344#endif 345 346 EnsureWebRtcAudioDeviceImpl(); 347 348 scoped_refptr<webrtc::PeerConnectionFactoryInterface> factory( 349 webrtc::CreatePeerConnectionFactory(worker_thread_, 350 signaling_thread_, 351 audio_device_.get(), 352 encoder_factory.release(), 353 decoder_factory.release())); 354 CHECK(factory.get()); 355 356 pc_factory_ = factory; 357 webrtc::PeerConnectionFactoryInterface::Options factory_options; 358 factory_options.disable_sctp_data_channels = false; 359 factory_options.disable_encryption = 360 cmd_line->HasSwitch(switches::kDisableWebRtcEncryption); 361 pc_factory_->SetOptions(factory_options); 362 363 // TODO(xians): Remove the following code after kDisableAudioTrackProcessing 364 // is removed. 365 if (!MediaStreamAudioProcessor::IsAudioTrackProcessingEnabled()) { 366 aec_dump_message_filter_ = AecDumpMessageFilter::Get(); 367 // In unit tests not creating a message filter, |aec_dump_message_filter_| 368 // will be NULL. We can just ignore that. Other unit tests and browser tests 369 // ensure that we do get the filter when we should. 370 if (aec_dump_message_filter_.get()) 371 aec_dump_message_filter_->AddDelegate(this); 372 } 373} 374 375bool PeerConnectionDependencyFactory::PeerConnectionFactoryCreated() { 376 return pc_factory_.get() != NULL; 377} 378 379scoped_refptr<webrtc::PeerConnectionInterface> 380PeerConnectionDependencyFactory::CreatePeerConnection( 381 const webrtc::PeerConnectionInterface::RTCConfiguration& config, 382 const webrtc::MediaConstraintsInterface* constraints, 383 blink::WebFrame* web_frame, 384 webrtc::PeerConnectionObserver* observer) { 385 CHECK(web_frame); 386 CHECK(observer); 387 if (!GetPcFactory().get()) 388 return NULL; 389 390 scoped_refptr<P2PPortAllocatorFactory> pa_factory = 391 new rtc::RefCountedObject<P2PPortAllocatorFactory>( 392 p2p_socket_dispatcher_.get(), 393 network_manager_, 394 socket_factory_.get(), 395 web_frame); 396 397 PeerConnectionIdentityService* identity_service = 398 new PeerConnectionIdentityService( 399 GURL(web_frame->document().url().spec()).GetOrigin()); 400 401 return GetPcFactory()->CreatePeerConnection(config, 402 constraints, 403 pa_factory.get(), 404 identity_service, 405 observer).get(); 406} 407 408scoped_refptr<webrtc::MediaStreamInterface> 409PeerConnectionDependencyFactory::CreateLocalMediaStream( 410 const std::string& label) { 411 return GetPcFactory()->CreateLocalMediaStream(label).get(); 412} 413 414scoped_refptr<webrtc::AudioSourceInterface> 415PeerConnectionDependencyFactory::CreateLocalAudioSource( 416 const webrtc::MediaConstraintsInterface* constraints) { 417 scoped_refptr<webrtc::AudioSourceInterface> source = 418 GetPcFactory()->CreateAudioSource(constraints).get(); 419 return source; 420} 421 422void PeerConnectionDependencyFactory::CreateLocalAudioTrack( 423 const blink::WebMediaStreamTrack& track) { 424 blink::WebMediaStreamSource source = track.source(); 425 DCHECK_EQ(source.type(), blink::WebMediaStreamSource::TypeAudio); 426 MediaStreamAudioSource* source_data = 427 static_cast<MediaStreamAudioSource*>(source.extraData()); 428 429 scoped_refptr<WebAudioCapturerSource> webaudio_source; 430 if (!source_data) { 431 if (source.requiresAudioConsumer()) { 432 // We're adding a WebAudio MediaStream. 433 // Create a specific capturer for each WebAudio consumer. 434 webaudio_source = CreateWebAudioSource(&source); 435 source_data = 436 static_cast<MediaStreamAudioSource*>(source.extraData()); 437 } else { 438 // TODO(perkj): Implement support for sources from 439 // remote MediaStreams. 440 NOTIMPLEMENTED(); 441 return; 442 } 443 } 444 445 // Creates an adapter to hold all the libjingle objects. 446 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( 447 WebRtcLocalAudioTrackAdapter::Create(track.id().utf8(), 448 source_data->local_audio_source())); 449 static_cast<webrtc::AudioTrackInterface*>(adapter.get())->set_enabled( 450 track.isEnabled()); 451 452 // TODO(xians): Merge |source| to the capturer(). We can't do this today 453 // because only one capturer() is supported while one |source| is created 454 // for each audio track. 455 scoped_ptr<WebRtcLocalAudioTrack> audio_track(new WebRtcLocalAudioTrack( 456 adapter.get(), source_data->GetAudioCapturer(), webaudio_source.get())); 457 458 StartLocalAudioTrack(audio_track.get()); 459 460 // Pass the ownership of the native local audio track to the blink track. 461 blink::WebMediaStreamTrack writable_track = track; 462 writable_track.setExtraData(audio_track.release()); 463} 464 465void PeerConnectionDependencyFactory::StartLocalAudioTrack( 466 WebRtcLocalAudioTrack* audio_track) { 467 // Add the WebRtcAudioDevice as the sink to the local audio track. 468 // TODO(xians): Remove the following line of code after the APM in WebRTC is 469 // completely deprecated. See http://crbug/365672. 470 if (!MediaStreamAudioProcessor::IsAudioTrackProcessingEnabled()) 471 audio_track->AddSink(GetWebRtcAudioDevice()); 472 473 // Start the audio track. This will hook the |audio_track| to the capturer 474 // as the sink of the audio, and only start the source of the capturer if 475 // it is the first audio track connecting to the capturer. 476 audio_track->Start(); 477} 478 479scoped_refptr<WebAudioCapturerSource> 480PeerConnectionDependencyFactory::CreateWebAudioSource( 481 blink::WebMediaStreamSource* source) { 482 DVLOG(1) << "PeerConnectionDependencyFactory::CreateWebAudioSource()"; 483 484 scoped_refptr<WebAudioCapturerSource> 485 webaudio_capturer_source(new WebAudioCapturerSource()); 486 MediaStreamAudioSource* source_data = new MediaStreamAudioSource(); 487 488 // Use the current default capturer for the WebAudio track so that the 489 // WebAudio track can pass a valid delay value and |need_audio_processing| 490 // flag to PeerConnection. 491 // TODO(xians): Remove this after moving APM to Chrome. 492 if (GetWebRtcAudioDevice()) { 493 source_data->SetAudioCapturer( 494 GetWebRtcAudioDevice()->GetDefaultCapturer()); 495 } 496 497 // Create a LocalAudioSource object which holds audio options. 498 // SetLocalAudioSource() affects core audio parts in third_party/Libjingle. 499 source_data->SetLocalAudioSource(CreateLocalAudioSource(NULL).get()); 500 source->setExtraData(source_data); 501 502 // Replace the default source with WebAudio as source instead. 503 source->addAudioConsumer(webaudio_capturer_source.get()); 504 505 return webaudio_capturer_source; 506} 507 508scoped_refptr<webrtc::VideoTrackInterface> 509PeerConnectionDependencyFactory::CreateLocalVideoTrack( 510 const std::string& id, 511 webrtc::VideoSourceInterface* source) { 512 return GetPcFactory()->CreateVideoTrack(id, source).get(); 513} 514 515scoped_refptr<webrtc::VideoTrackInterface> 516PeerConnectionDependencyFactory::CreateLocalVideoTrack( 517 const std::string& id, cricket::VideoCapturer* capturer) { 518 if (!capturer) { 519 LOG(ERROR) << "CreateLocalVideoTrack called with null VideoCapturer."; 520 return NULL; 521 } 522 523 // Create video source from the |capturer|. 524 scoped_refptr<webrtc::VideoSourceInterface> source = 525 GetPcFactory()->CreateVideoSource(capturer, NULL).get(); 526 527 // Create native track from the source. 528 return GetPcFactory()->CreateVideoTrack(id, source.get()).get(); 529} 530 531webrtc::SessionDescriptionInterface* 532PeerConnectionDependencyFactory::CreateSessionDescription( 533 const std::string& type, 534 const std::string& sdp, 535 webrtc::SdpParseError* error) { 536 return webrtc::CreateSessionDescription(type, sdp, error); 537} 538 539webrtc::IceCandidateInterface* 540PeerConnectionDependencyFactory::CreateIceCandidate( 541 const std::string& sdp_mid, 542 int sdp_mline_index, 543 const std::string& sdp) { 544 return webrtc::CreateIceCandidate(sdp_mid, sdp_mline_index, sdp); 545} 546 547WebRtcAudioDeviceImpl* 548PeerConnectionDependencyFactory::GetWebRtcAudioDevice() { 549 return audio_device_.get(); 550} 551 552void PeerConnectionDependencyFactory::InitializeWorkerThread( 553 rtc::Thread** thread, 554 base::WaitableEvent* event) { 555 jingle_glue::JingleThreadWrapper::EnsureForCurrentMessageLoop(); 556 jingle_glue::JingleThreadWrapper::current()->set_send_allowed(true); 557 *thread = jingle_glue::JingleThreadWrapper::current(); 558 event->Signal(); 559} 560 561void PeerConnectionDependencyFactory::CreateIpcNetworkManagerOnWorkerThread( 562 base::WaitableEvent* event) { 563 DCHECK_EQ(base::MessageLoop::current(), chrome_worker_thread_.message_loop()); 564 network_manager_ = new IpcNetworkManager(p2p_socket_dispatcher_.get()); 565 event->Signal(); 566} 567 568void PeerConnectionDependencyFactory::DeleteIpcNetworkManager() { 569 DCHECK_EQ(base::MessageLoop::current(), chrome_worker_thread_.message_loop()); 570 delete network_manager_; 571 network_manager_ = NULL; 572} 573 574void PeerConnectionDependencyFactory::CleanupPeerConnectionFactory() { 575 pc_factory_ = NULL; 576 if (network_manager_) { 577 // The network manager needs to free its resources on the thread they were 578 // created, which is the worked thread. 579 if (chrome_worker_thread_.IsRunning()) { 580 chrome_worker_thread_.message_loop()->PostTask(FROM_HERE, base::Bind( 581 &PeerConnectionDependencyFactory::DeleteIpcNetworkManager, 582 base::Unretained(this))); 583 // Stopping the thread will wait until all tasks have been 584 // processed before returning. We wait for the above task to finish before 585 // letting the the function continue to avoid any potential race issues. 586 chrome_worker_thread_.Stop(); 587 } else { 588 NOTREACHED() << "Worker thread not running."; 589 } 590 } 591} 592 593scoped_refptr<WebRtcAudioCapturer> 594PeerConnectionDependencyFactory::CreateAudioCapturer( 595 int render_view_id, 596 const StreamDeviceInfo& device_info, 597 const blink::WebMediaConstraints& constraints, 598 MediaStreamAudioSource* audio_source) { 599 // TODO(xians): Handle the cases when gUM is called without a proper render 600 // view, for example, by an extension. 601 DCHECK_GE(render_view_id, 0); 602 603 EnsureWebRtcAudioDeviceImpl(); 604 DCHECK(GetWebRtcAudioDevice()); 605 return WebRtcAudioCapturer::CreateCapturer(render_view_id, device_info, 606 constraints, 607 GetWebRtcAudioDevice(), 608 audio_source); 609} 610 611void PeerConnectionDependencyFactory::AddNativeAudioTrackToBlinkTrack( 612 webrtc::MediaStreamTrackInterface* native_track, 613 const blink::WebMediaStreamTrack& webkit_track, 614 bool is_local_track) { 615 DCHECK(!webkit_track.isNull() && !webkit_track.extraData()); 616 DCHECK_EQ(blink::WebMediaStreamSource::TypeAudio, 617 webkit_track.source().type()); 618 blink::WebMediaStreamTrack track = webkit_track; 619 620 DVLOG(1) << "AddNativeTrackToBlinkTrack() audio"; 621 track.setExtraData( 622 new MediaStreamTrack( 623 static_cast<webrtc::AudioTrackInterface*>(native_track), 624 is_local_track)); 625} 626 627scoped_refptr<base::MessageLoopProxy> 628PeerConnectionDependencyFactory::GetWebRtcWorkerThread() const { 629 DCHECK(CalledOnValidThread()); 630 return chrome_worker_thread_.message_loop_proxy(); 631} 632 633void PeerConnectionDependencyFactory::OnAecDumpFile( 634 const IPC::PlatformFileForTransit& file_handle) { 635 DCHECK(CalledOnValidThread()); 636 DCHECK(!MediaStreamAudioProcessor::IsAudioTrackProcessingEnabled()); 637 DCHECK(PeerConnectionFactoryCreated()); 638 639 base::File file = IPC::PlatformFileForTransitToFile(file_handle); 640 DCHECK(file.IsValid()); 641 642 // |pc_factory_| always takes ownership of |aec_dump_file|. If StartAecDump() 643 // fails, |aec_dump_file| will be closed. 644 if (!GetPcFactory()->StartAecDump(file.TakePlatformFile())) 645 VLOG(1) << "Could not start AEC dump."; 646} 647 648void PeerConnectionDependencyFactory::OnDisableAecDump() { 649 DCHECK(CalledOnValidThread()); 650 DCHECK(!MediaStreamAudioProcessor::IsAudioTrackProcessingEnabled()); 651 // Do nothing. We never disable AEC dump for non-track-processing case. 652} 653 654void PeerConnectionDependencyFactory::OnIpcClosing() { 655 DCHECK(CalledOnValidThread()); 656 aec_dump_message_filter_ = NULL; 657} 658 659void PeerConnectionDependencyFactory::EnsureWebRtcAudioDeviceImpl() { 660 if (audio_device_.get()) 661 return; 662 663 audio_device_ = new WebRtcAudioDeviceImpl(); 664} 665 666} // namespace content 667