| // Copyright 2014 The Chromium Authors. All rights reserved. |
| // Use of this source code is governed by a BSD-style license that can be |
| // found in the LICENSE file. |
| |
| #include "content/renderer/media/webrtc/peer_connection_dependency_factory.h" |
| |
| #include <vector> |
| |
| #include "base/command_line.h" |
| #include "base/strings/utf_string_conversions.h" |
| #include "base/synchronization/waitable_event.h" |
| #include "content/common/media/media_stream_messages.h" |
| #include "content/public/common/content_switches.h" |
| #include "content/public/common/renderer_preferences.h" |
| #include "content/renderer/media/media_stream.h" |
| #include "content/renderer/media/media_stream_audio_processor.h" |
| #include "content/renderer/media/media_stream_audio_processor_options.h" |
| #include "content/renderer/media/media_stream_audio_source.h" |
| #include "content/renderer/media/media_stream_video_source.h" |
| #include "content/renderer/media/media_stream_video_track.h" |
| #include "content/renderer/media/peer_connection_identity_service.h" |
| #include "content/renderer/media/rtc_media_constraints.h" |
| #include "content/renderer/media/rtc_peer_connection_handler.h" |
| #include "content/renderer/media/rtc_video_decoder_factory.h" |
| #include "content/renderer/media/rtc_video_encoder_factory.h" |
| #include "content/renderer/media/webaudio_capturer_source.h" |
| #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h" |
| #include "content/renderer/media/webrtc/webrtc_video_capturer_adapter.h" |
| #include "content/renderer/media/webrtc_audio_device_impl.h" |
| #include "content/renderer/media/webrtc_local_audio_track.h" |
| #include "content/renderer/media/webrtc_logging.h" |
| #include "content/renderer/media/webrtc_uma_histograms.h" |
| #include "content/renderer/p2p/ipc_network_manager.h" |
| #include "content/renderer/p2p/ipc_socket_factory.h" |
| #include "content/renderer/p2p/port_allocator.h" |
| #include "content/renderer/render_thread_impl.h" |
| #include "content/renderer/render_view_impl.h" |
| #include "jingle/glue/thread_wrapper.h" |
| #include "media/filters/gpu_video_accelerator_factories.h" |
| #include "third_party/WebKit/public/platform/WebMediaConstraints.h" |
| #include "third_party/WebKit/public/platform/WebMediaStream.h" |
| #include "third_party/WebKit/public/platform/WebMediaStreamSource.h" |
| #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h" |
| #include "third_party/WebKit/public/platform/WebURL.h" |
| #include "third_party/WebKit/public/web/WebDocument.h" |
| #include "third_party/WebKit/public/web/WebFrame.h" |
| #include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface.h" |
| |
| #if defined(USE_OPENSSL) |
| #include "third_party/webrtc/base/ssladapter.h" |
| #else |
| #include "net/socket/nss_ssl_util.h" |
| #endif |
| |
| #if defined(OS_ANDROID) |
| #include "media/base/android/media_codec_bridge.h" |
| #endif |
| |
| namespace content { |
| |
| // Map of corresponding media constraints and platform effects. |
| struct { |
| const char* constraint; |
| const media::AudioParameters::PlatformEffectsMask effect; |
| } const kConstraintEffectMap[] = { |
| { content::kMediaStreamAudioDucking, |
| media::AudioParameters::DUCKING }, |
| { webrtc::MediaConstraintsInterface::kEchoCancellation, |
| media::AudioParameters::ECHO_CANCELLER }, |
| }; |
| |
| // If any platform effects are available, check them against the constraints. |
| // Disable effects to match false constraints, but if a constraint is true, set |
| // the constraint to false to later disable the software effect. |
| // |
| // This function may modify both |constraints| and |effects|. |
| void HarmonizeConstraintsAndEffects(RTCMediaConstraints* constraints, |
| int* effects) { |
| if (*effects != media::AudioParameters::NO_EFFECTS) { |
| for (size_t i = 0; i < arraysize(kConstraintEffectMap); ++i) { |
| bool value; |
| size_t is_mandatory = 0; |
| if (!webrtc::FindConstraint(constraints, |
| kConstraintEffectMap[i].constraint, |
| &value, |
| &is_mandatory) || !value) { |
| // If the constraint is false, or does not exist, disable the platform |
| // effect. |
| *effects &= ~kConstraintEffectMap[i].effect; |
| DVLOG(1) << "Disabling platform effect: " |
| << kConstraintEffectMap[i].effect; |
| } else if (*effects & kConstraintEffectMap[i].effect) { |
| // If the constraint is true, leave the platform effect enabled, and |
| // set the constraint to false to later disable the software effect. |
| if (is_mandatory) { |
| constraints->AddMandatory(kConstraintEffectMap[i].constraint, |
| webrtc::MediaConstraintsInterface::kValueFalse, true); |
| } else { |
| constraints->AddOptional(kConstraintEffectMap[i].constraint, |
| webrtc::MediaConstraintsInterface::kValueFalse, true); |
| } |
| DVLOG(1) << "Disabling constraint: " |
| << kConstraintEffectMap[i].constraint; |
| } else if (kConstraintEffectMap[i].effect == |
| media::AudioParameters::DUCKING && value && !is_mandatory) { |
| // Special handling of the DUCKING flag that sets the optional |
| // constraint to |false| to match what the device will support. |
| constraints->AddOptional(kConstraintEffectMap[i].constraint, |
| webrtc::MediaConstraintsInterface::kValueFalse, true); |
| // No need to modify |effects| since the ducking flag is already off. |
| DCHECK((*effects & media::AudioParameters::DUCKING) == 0); |
| } |
| } |
| } |
| } |
| |
| class P2PPortAllocatorFactory : public webrtc::PortAllocatorFactoryInterface { |
| public: |
| P2PPortAllocatorFactory(P2PSocketDispatcher* socket_dispatcher, |
| rtc::NetworkManager* network_manager, |
| rtc::PacketSocketFactory* socket_factory, |
| bool enable_multiple_routes) |
| : socket_dispatcher_(socket_dispatcher), |
| network_manager_(network_manager), |
| socket_factory_(socket_factory), |
| enable_multiple_routes_(enable_multiple_routes) {} |
| |
| cricket::PortAllocator* CreatePortAllocator( |
| const std::vector<StunConfiguration>& stun_servers, |
| const std::vector<TurnConfiguration>& turn_configurations) override { |
| P2PPortAllocator::Config config; |
| for (size_t i = 0; i < stun_servers.size(); ++i) { |
| config.stun_servers.insert(rtc::SocketAddress( |
| stun_servers[i].server.hostname(), |
| stun_servers[i].server.port())); |
| } |
| for (size_t i = 0; i < turn_configurations.size(); ++i) { |
| P2PPortAllocator::Config::RelayServerConfig relay_config; |
| relay_config.server_address = turn_configurations[i].server.hostname(); |
| relay_config.port = turn_configurations[i].server.port(); |
| relay_config.username = turn_configurations[i].username; |
| relay_config.password = turn_configurations[i].password; |
| relay_config.transport_type = turn_configurations[i].transport_type; |
| relay_config.secure = turn_configurations[i].secure; |
| config.relays.push_back(relay_config); |
| |
| // Use turn servers as stun servers. |
| config.stun_servers.insert(rtc::SocketAddress( |
| turn_configurations[i].server.hostname(), |
| turn_configurations[i].server.port())); |
| } |
| config.enable_multiple_routes = enable_multiple_routes_; |
| |
| return new P2PPortAllocator( |
| socket_dispatcher_.get(), network_manager_, socket_factory_, config); |
| } |
| |
| protected: |
| ~P2PPortAllocatorFactory() override {} |
| |
| private: |
| scoped_refptr<P2PSocketDispatcher> socket_dispatcher_; |
| // |network_manager_| and |socket_factory_| are a weak references, owned by |
| // PeerConnectionDependencyFactory. |
| rtc::NetworkManager* network_manager_; |
| rtc::PacketSocketFactory* socket_factory_; |
| |
| // When false, only 'any' address (all 0s) will be bound for address |
| // discovery. |
| bool enable_multiple_routes_; |
| }; |
| |
| PeerConnectionDependencyFactory::PeerConnectionDependencyFactory( |
| P2PSocketDispatcher* p2p_socket_dispatcher) |
| : network_manager_(NULL), |
| p2p_socket_dispatcher_(p2p_socket_dispatcher), |
| signaling_thread_(NULL), |
| worker_thread_(NULL), |
| chrome_signaling_thread_("Chrome_libJingle_Signaling"), |
| chrome_worker_thread_("Chrome_libJingle_WorkerThread") { |
| } |
| |
| PeerConnectionDependencyFactory::~PeerConnectionDependencyFactory() { |
| DVLOG(1) << "~PeerConnectionDependencyFactory()"; |
| DCHECK(pc_factory_ == NULL); |
| } |
| |
| blink::WebRTCPeerConnectionHandler* |
| PeerConnectionDependencyFactory::CreateRTCPeerConnectionHandler( |
| blink::WebRTCPeerConnectionHandlerClient* client) { |
| // Save histogram data so we can see how much PeerConnetion is used. |
| // The histogram counts the number of calls to the JS API |
| // webKitRTCPeerConnection. |
| UpdateWebRTCMethodCount(WEBKIT_RTC_PEER_CONNECTION); |
| |
| return new RTCPeerConnectionHandler(client, this); |
| } |
| |
| bool PeerConnectionDependencyFactory::InitializeMediaStreamAudioSource( |
| int render_view_id, |
| const blink::WebMediaConstraints& audio_constraints, |
| MediaStreamAudioSource* source_data) { |
| DVLOG(1) << "InitializeMediaStreamAudioSources()"; |
| |
| // Do additional source initialization if the audio source is a valid |
| // microphone or tab audio. |
| RTCMediaConstraints native_audio_constraints(audio_constraints); |
| MediaAudioConstraints::ApplyFixedAudioConstraints(&native_audio_constraints); |
| |
| StreamDeviceInfo device_info = source_data->device_info(); |
| RTCMediaConstraints constraints = native_audio_constraints; |
| // May modify both |constraints| and |effects|. |
| HarmonizeConstraintsAndEffects(&constraints, |
| &device_info.device.input.effects); |
| |
| scoped_refptr<WebRtcAudioCapturer> capturer( |
| CreateAudioCapturer(render_view_id, device_info, audio_constraints, |
| source_data)); |
| if (!capturer.get()) { |
| const std::string log_string = |
| "PCDF::InitializeMediaStreamAudioSource: fails to create capturer"; |
| WebRtcLogMessage(log_string); |
| DVLOG(1) << log_string; |
| // TODO(xians): Don't we need to check if source_observer is observing |
| // something? If not, then it looks like we have a leak here. |
| // OTOH, if it _is_ observing something, then the callback might |
| // be called multiple times which is likely also a bug. |
| return false; |
| } |
| source_data->SetAudioCapturer(capturer.get()); |
| |
| // Creates a LocalAudioSource object which holds audio options. |
| // TODO(xians): The option should apply to the track instead of the source. |
| // TODO(perkj): Move audio constraints parsing to Chrome. |
| // Currently there are a few constraints that are parsed by libjingle and |
| // the state is set to ended if parsing fails. |
| scoped_refptr<webrtc::AudioSourceInterface> rtc_source( |
| CreateLocalAudioSource(&constraints).get()); |
| if (rtc_source->state() != webrtc::MediaSourceInterface::kLive) { |
| DLOG(WARNING) << "Failed to create rtc LocalAudioSource."; |
| return false; |
| } |
| source_data->SetLocalAudioSource(rtc_source.get()); |
| return true; |
| } |
| |
| WebRtcVideoCapturerAdapter* |
| PeerConnectionDependencyFactory::CreateVideoCapturer( |
| bool is_screeencast) { |
| // We need to make sure the libjingle thread wrappers have been created |
| // before we can use an instance of a WebRtcVideoCapturerAdapter. This is |
| // since the base class of WebRtcVideoCapturerAdapter is a |
| // cricket::VideoCapturer and it uses the libjingle thread wrappers. |
| if (!GetPcFactory().get()) |
| return NULL; |
| return new WebRtcVideoCapturerAdapter(is_screeencast); |
| } |
| |
| scoped_refptr<webrtc::VideoSourceInterface> |
| PeerConnectionDependencyFactory::CreateVideoSource( |
| cricket::VideoCapturer* capturer, |
| const blink::WebMediaConstraints& constraints) { |
| RTCMediaConstraints webrtc_constraints(constraints); |
| scoped_refptr<webrtc::VideoSourceInterface> source = |
| GetPcFactory()->CreateVideoSource(capturer, &webrtc_constraints).get(); |
| return source; |
| } |
| |
| const scoped_refptr<webrtc::PeerConnectionFactoryInterface>& |
| PeerConnectionDependencyFactory::GetPcFactory() { |
| if (!pc_factory_.get()) |
| CreatePeerConnectionFactory(); |
| CHECK(pc_factory_.get()); |
| return pc_factory_; |
| } |
| |
| |
| void PeerConnectionDependencyFactory::WillDestroyCurrentMessageLoop() { |
| CleanupPeerConnectionFactory(); |
| } |
| |
| void PeerConnectionDependencyFactory::CreatePeerConnectionFactory() { |
| DCHECK(!pc_factory_.get()); |
| DCHECK(!signaling_thread_); |
| DCHECK(!worker_thread_); |
| DCHECK(!network_manager_); |
| DCHECK(!socket_factory_); |
| DCHECK(!chrome_signaling_thread_.IsRunning()); |
| DCHECK(!chrome_worker_thread_.IsRunning()); |
| |
| DVLOG(1) << "PeerConnectionDependencyFactory::CreatePeerConnectionFactory()"; |
| |
| base::MessageLoop::current()->AddDestructionObserver(this); |
| // To allow sending to the signaling/worker threads. |
| jingle_glue::JingleThreadWrapper::EnsureForCurrentMessageLoop(); |
| jingle_glue::JingleThreadWrapper::current()->set_send_allowed(true); |
| |
| CHECK(chrome_signaling_thread_.Start()); |
| CHECK(chrome_worker_thread_.Start()); |
| |
| base::WaitableEvent start_worker_event(true, false); |
| chrome_worker_thread_.message_loop()->PostTask(FROM_HERE, base::Bind( |
| &PeerConnectionDependencyFactory::InitializeWorkerThread, |
| base::Unretained(this), |
| &worker_thread_, |
| &start_worker_event)); |
| |
| base::WaitableEvent create_network_manager_event(true, false); |
| chrome_worker_thread_.message_loop()->PostTask(FROM_HERE, base::Bind( |
| &PeerConnectionDependencyFactory::CreateIpcNetworkManagerOnWorkerThread, |
| base::Unretained(this), |
| &create_network_manager_event)); |
| |
| start_worker_event.Wait(); |
| create_network_manager_event.Wait(); |
| |
| CHECK(worker_thread_); |
| |
| // Init SSL, which will be needed by PeerConnection. |
| #if defined(USE_OPENSSL) |
| if (!rtc::InitializeSSL()) { |
| LOG(ERROR) << "Failed on InitializeSSL."; |
| NOTREACHED(); |
| return; |
| } |
| #else |
| // TODO(ronghuawu): Replace this call with InitializeSSL. |
| net::EnsureNSSSSLInit(); |
| #endif |
| |
| base::WaitableEvent start_signaling_event(true, false); |
| chrome_signaling_thread_.message_loop()->PostTask(FROM_HERE, base::Bind( |
| &PeerConnectionDependencyFactory::InitializeSignalingThread, |
| base::Unretained(this), |
| RenderThreadImpl::current()->GetGpuFactories(), |
| &start_signaling_event)); |
| |
| start_signaling_event.Wait(); |
| CHECK(signaling_thread_); |
| } |
| |
| void PeerConnectionDependencyFactory::InitializeSignalingThread( |
| const scoped_refptr<media::GpuVideoAcceleratorFactories>& gpu_factories, |
| base::WaitableEvent* event) { |
| DCHECK(chrome_signaling_thread_.task_runner()->BelongsToCurrentThread()); |
| DCHECK(worker_thread_); |
| DCHECK(p2p_socket_dispatcher_.get()); |
| |
| jingle_glue::JingleThreadWrapper::EnsureForCurrentMessageLoop(); |
| jingle_glue::JingleThreadWrapper::current()->set_send_allowed(true); |
| signaling_thread_ = jingle_glue::JingleThreadWrapper::current(); |
| |
| EnsureWebRtcAudioDeviceImpl(); |
| |
| socket_factory_.reset( |
| new IpcPacketSocketFactory(p2p_socket_dispatcher_.get())); |
| |
| scoped_ptr<cricket::WebRtcVideoDecoderFactory> decoder_factory; |
| scoped_ptr<cricket::WebRtcVideoEncoderFactory> encoder_factory; |
| |
| const base::CommandLine* cmd_line = base::CommandLine::ForCurrentProcess(); |
| if (gpu_factories.get()) { |
| if (!cmd_line->HasSwitch(switches::kDisableWebRtcHWDecoding)) |
| decoder_factory.reset(new RTCVideoDecoderFactory(gpu_factories)); |
| |
| if (!cmd_line->HasSwitch(switches::kDisableWebRtcHWEncoding)) |
| encoder_factory.reset(new RTCVideoEncoderFactory(gpu_factories)); |
| } |
| |
| #if defined(OS_ANDROID) |
| if (!media::MediaCodecBridge::SupportsSetParameters()) |
| encoder_factory.reset(); |
| #endif |
| |
| pc_factory_ = webrtc::CreatePeerConnectionFactory( |
| worker_thread_, signaling_thread_, audio_device_.get(), |
| encoder_factory.release(), decoder_factory.release()); |
| CHECK(pc_factory_.get()); |
| |
| webrtc::PeerConnectionFactoryInterface::Options factory_options; |
| factory_options.disable_sctp_data_channels = false; |
| factory_options.disable_encryption = |
| cmd_line->HasSwitch(switches::kDisableWebRtcEncryption); |
| pc_factory_->SetOptions(factory_options); |
| |
| event->Signal(); |
| } |
| |
| bool PeerConnectionDependencyFactory::PeerConnectionFactoryCreated() { |
| return pc_factory_.get() != NULL; |
| } |
| |
| scoped_refptr<webrtc::PeerConnectionInterface> |
| PeerConnectionDependencyFactory::CreatePeerConnection( |
| const webrtc::PeerConnectionInterface::RTCConfiguration& config, |
| const webrtc::MediaConstraintsInterface* constraints, |
| blink::WebFrame* web_frame, |
| webrtc::PeerConnectionObserver* observer) { |
| CHECK(web_frame); |
| CHECK(observer); |
| if (!GetPcFactory().get()) |
| return NULL; |
| |
| // Copy the flag from Preference associated with this WebFrame. |
| bool enable_multiple_routes = true; |
| if (web_frame && web_frame->view()) { |
| RenderViewImpl* renderer_view_impl = |
| RenderViewImpl::FromWebView(web_frame->view()); |
| if (renderer_view_impl) { |
| enable_multiple_routes = renderer_view_impl->renderer_preferences() |
| .enable_webrtc_multiple_routes; |
| } |
| } |
| |
| scoped_refptr<P2PPortAllocatorFactory> pa_factory = |
| new rtc::RefCountedObject<P2PPortAllocatorFactory>( |
| p2p_socket_dispatcher_.get(), network_manager_, socket_factory_.get(), |
| enable_multiple_routes); |
| |
| PeerConnectionIdentityService* identity_service = |
| new PeerConnectionIdentityService( |
| GURL(web_frame->document().url().spec()).GetOrigin()); |
| |
| return GetPcFactory()->CreatePeerConnection(config, |
| constraints, |
| pa_factory.get(), |
| identity_service, |
| observer).get(); |
| } |
| |
| scoped_refptr<webrtc::MediaStreamInterface> |
| PeerConnectionDependencyFactory::CreateLocalMediaStream( |
| const std::string& label) { |
| return GetPcFactory()->CreateLocalMediaStream(label).get(); |
| } |
| |
| scoped_refptr<webrtc::AudioSourceInterface> |
| PeerConnectionDependencyFactory::CreateLocalAudioSource( |
| const webrtc::MediaConstraintsInterface* constraints) { |
| scoped_refptr<webrtc::AudioSourceInterface> source = |
| GetPcFactory()->CreateAudioSource(constraints).get(); |
| return source; |
| } |
| |
| void PeerConnectionDependencyFactory::CreateLocalAudioTrack( |
| const blink::WebMediaStreamTrack& track) { |
| blink::WebMediaStreamSource source = track.source(); |
| DCHECK_EQ(source.type(), blink::WebMediaStreamSource::TypeAudio); |
| MediaStreamAudioSource* source_data = |
| static_cast<MediaStreamAudioSource*>(source.extraData()); |
| |
| scoped_refptr<WebAudioCapturerSource> webaudio_source; |
| if (!source_data) { |
| if (source.requiresAudioConsumer()) { |
| // We're adding a WebAudio MediaStream. |
| // Create a specific capturer for each WebAudio consumer. |
| webaudio_source = CreateWebAudioSource(&source); |
| source_data = |
| static_cast<MediaStreamAudioSource*>(source.extraData()); |
| } else { |
| // TODO(perkj): Implement support for sources from |
| // remote MediaStreams. |
| NOTIMPLEMENTED(); |
| return; |
| } |
| } |
| |
| // Creates an adapter to hold all the libjingle objects. |
| scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( |
| WebRtcLocalAudioTrackAdapter::Create(track.id().utf8(), |
| source_data->local_audio_source())); |
| static_cast<webrtc::AudioTrackInterface*>(adapter.get())->set_enabled( |
| track.isEnabled()); |
| |
| // TODO(xians): Merge |source| to the capturer(). We can't do this today |
| // because only one capturer() is supported while one |source| is created |
| // for each audio track. |
| scoped_ptr<WebRtcLocalAudioTrack> audio_track(new WebRtcLocalAudioTrack( |
| adapter.get(), source_data->GetAudioCapturer(), webaudio_source.get())); |
| |
| StartLocalAudioTrack(audio_track.get()); |
| |
| // Pass the ownership of the native local audio track to the blink track. |
| blink::WebMediaStreamTrack writable_track = track; |
| writable_track.setExtraData(audio_track.release()); |
| } |
| |
| void PeerConnectionDependencyFactory::StartLocalAudioTrack( |
| WebRtcLocalAudioTrack* audio_track) { |
| // Start the audio track. This will hook the |audio_track| to the capturer |
| // as the sink of the audio, and only start the source of the capturer if |
| // it is the first audio track connecting to the capturer. |
| audio_track->Start(); |
| } |
| |
| scoped_refptr<WebAudioCapturerSource> |
| PeerConnectionDependencyFactory::CreateWebAudioSource( |
| blink::WebMediaStreamSource* source) { |
| DVLOG(1) << "PeerConnectionDependencyFactory::CreateWebAudioSource()"; |
| |
| scoped_refptr<WebAudioCapturerSource> |
| webaudio_capturer_source(new WebAudioCapturerSource()); |
| MediaStreamAudioSource* source_data = new MediaStreamAudioSource(); |
| |
| // Use the current default capturer for the WebAudio track so that the |
| // WebAudio track can pass a valid delay value and |need_audio_processing| |
| // flag to PeerConnection. |
| // TODO(xians): Remove this after moving APM to Chrome. |
| if (GetWebRtcAudioDevice()) { |
| source_data->SetAudioCapturer( |
| GetWebRtcAudioDevice()->GetDefaultCapturer()); |
| } |
| |
| // Create a LocalAudioSource object which holds audio options. |
| // SetLocalAudioSource() affects core audio parts in third_party/Libjingle. |
| source_data->SetLocalAudioSource(CreateLocalAudioSource(NULL).get()); |
| source->setExtraData(source_data); |
| |
| // Replace the default source with WebAudio as source instead. |
| source->addAudioConsumer(webaudio_capturer_source.get()); |
| |
| return webaudio_capturer_source; |
| } |
| |
| scoped_refptr<webrtc::VideoTrackInterface> |
| PeerConnectionDependencyFactory::CreateLocalVideoTrack( |
| const std::string& id, |
| webrtc::VideoSourceInterface* source) { |
| return GetPcFactory()->CreateVideoTrack(id, source).get(); |
| } |
| |
| scoped_refptr<webrtc::VideoTrackInterface> |
| PeerConnectionDependencyFactory::CreateLocalVideoTrack( |
| const std::string& id, cricket::VideoCapturer* capturer) { |
| if (!capturer) { |
| LOG(ERROR) << "CreateLocalVideoTrack called with null VideoCapturer."; |
| return NULL; |
| } |
| |
| // Create video source from the |capturer|. |
| scoped_refptr<webrtc::VideoSourceInterface> source = |
| GetPcFactory()->CreateVideoSource(capturer, NULL).get(); |
| |
| // Create native track from the source. |
| return GetPcFactory()->CreateVideoTrack(id, source.get()).get(); |
| } |
| |
| webrtc::SessionDescriptionInterface* |
| PeerConnectionDependencyFactory::CreateSessionDescription( |
| const std::string& type, |
| const std::string& sdp, |
| webrtc::SdpParseError* error) { |
| return webrtc::CreateSessionDescription(type, sdp, error); |
| } |
| |
| webrtc::IceCandidateInterface* |
| PeerConnectionDependencyFactory::CreateIceCandidate( |
| const std::string& sdp_mid, |
| int sdp_mline_index, |
| const std::string& sdp) { |
| return webrtc::CreateIceCandidate(sdp_mid, sdp_mline_index, sdp); |
| } |
| |
| WebRtcAudioDeviceImpl* |
| PeerConnectionDependencyFactory::GetWebRtcAudioDevice() { |
| return audio_device_.get(); |
| } |
| |
| void PeerConnectionDependencyFactory::InitializeWorkerThread( |
| rtc::Thread** thread, |
| base::WaitableEvent* event) { |
| jingle_glue::JingleThreadWrapper::EnsureForCurrentMessageLoop(); |
| jingle_glue::JingleThreadWrapper::current()->set_send_allowed(true); |
| *thread = jingle_glue::JingleThreadWrapper::current(); |
| event->Signal(); |
| } |
| |
| void PeerConnectionDependencyFactory::CreateIpcNetworkManagerOnWorkerThread( |
| base::WaitableEvent* event) { |
| DCHECK_EQ(base::MessageLoop::current(), chrome_worker_thread_.message_loop()); |
| network_manager_ = new IpcNetworkManager(p2p_socket_dispatcher_.get()); |
| event->Signal(); |
| } |
| |
| void PeerConnectionDependencyFactory::DeleteIpcNetworkManager() { |
| DCHECK_EQ(base::MessageLoop::current(), chrome_worker_thread_.message_loop()); |
| delete network_manager_; |
| network_manager_ = NULL; |
| } |
| |
| void PeerConnectionDependencyFactory::CleanupPeerConnectionFactory() { |
| DVLOG(1) << "PeerConnectionDependencyFactory::CleanupPeerConnectionFactory()"; |
| pc_factory_ = NULL; |
| if (network_manager_) { |
| // The network manager needs to free its resources on the thread they were |
| // created, which is the worked thread. |
| if (chrome_worker_thread_.IsRunning()) { |
| chrome_worker_thread_.message_loop()->PostTask(FROM_HERE, base::Bind( |
| &PeerConnectionDependencyFactory::DeleteIpcNetworkManager, |
| base::Unretained(this))); |
| // Stopping the thread will wait until all tasks have been |
| // processed before returning. We wait for the above task to finish before |
| // letting the the function continue to avoid any potential race issues. |
| chrome_worker_thread_.Stop(); |
| } else { |
| NOTREACHED() << "Worker thread not running."; |
| } |
| } |
| } |
| |
| scoped_refptr<WebRtcAudioCapturer> |
| PeerConnectionDependencyFactory::CreateAudioCapturer( |
| int render_view_id, |
| const StreamDeviceInfo& device_info, |
| const blink::WebMediaConstraints& constraints, |
| MediaStreamAudioSource* audio_source) { |
| // TODO(xians): Handle the cases when gUM is called without a proper render |
| // view, for example, by an extension. |
| DCHECK_GE(render_view_id, 0); |
| |
| EnsureWebRtcAudioDeviceImpl(); |
| DCHECK(GetWebRtcAudioDevice()); |
| return WebRtcAudioCapturer::CreateCapturer(render_view_id, device_info, |
| constraints, |
| GetWebRtcAudioDevice(), |
| audio_source); |
| } |
| |
| scoped_refptr<base::MessageLoopProxy> |
| PeerConnectionDependencyFactory::GetWebRtcWorkerThread() const { |
| DCHECK(CalledOnValidThread()); |
| return chrome_worker_thread_.message_loop_proxy(); |
| } |
| |
| scoped_refptr<base::MessageLoopProxy> |
| PeerConnectionDependencyFactory::GetWebRtcSignalingThread() const { |
| DCHECK(CalledOnValidThread()); |
| return chrome_signaling_thread_.message_loop_proxy(); |
| } |
| |
| void PeerConnectionDependencyFactory::EnsureWebRtcAudioDeviceImpl() { |
| if (audio_device_.get()) |
| return; |
| |
| audio_device_ = new WebRtcAudioDeviceImpl(); |
| } |
| |
| } // namespace content |