| // 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 <stddef.h> |
| |
| #include <memory> |
| #include <utility> |
| #include <vector> |
| |
| #include "base/bind.h" |
| #include "base/bind_helpers.h" |
| #include "base/command_line.h" |
| #include "base/location.h" |
| #include "base/logging.h" |
| #include "base/macros.h" |
| #include "base/metrics/field_trial.h" |
| #include "base/strings/string_util.h" |
| #include "base/strings/utf_string_conversions.h" |
| #include "base/synchronization/waitable_event.h" |
| #include "base/threading/thread_task_runner_handle.h" |
| #include "build/build_config.h" |
| #include "content/public/common/buildflags.h" |
| #include "content/public/common/content_client.h" |
| #include "content/public/common/content_features.h" |
| #include "content/public/common/content_switches.h" |
| #include "content/public/common/feature_h264_with_openh264_ffmpeg.h" |
| #include "content/public/common/renderer_preferences.h" |
| #include "content/public/common/webrtc_ip_handling_policy.h" |
| #include "content/public/renderer/content_renderer_client.h" |
| #include "content/renderer/media/stream/media_stream_video_source.h" |
| #include "content/renderer/media/stream/media_stream_video_track.h" |
| #include "content/renderer/media/webrtc/audio_codec_factory.h" |
| #include "content/renderer/media/webrtc/rtc_peer_connection_handler.h" |
| #include "content/renderer/media/webrtc/stun_field_trial.h" |
| #include "content/renderer/media/webrtc/video_codec_factory.h" |
| #include "content/renderer/media/webrtc/webrtc_audio_device_impl.h" |
| #include "content/renderer/media/webrtc/webrtc_uma_histograms.h" |
| #include "content/renderer/media/webrtc_logging.h" |
| #include "content/renderer/p2p/empty_network_manager.h" |
| #include "content/renderer/p2p/filtering_network_manager.h" |
| #include "content/renderer/p2p/ipc_network_manager.h" |
| #include "content/renderer/p2p/ipc_socket_factory.h" |
| #include "content/renderer/p2p/mdns_responder_adapter.h" |
| #include "content/renderer/p2p/port_allocator.h" |
| #include "content/renderer/render_frame_impl.h" |
| #include "content/renderer/render_thread_impl.h" |
| #include "content/renderer/render_view_impl.h" |
| #include "crypto/openssl_util.h" |
| #include "jingle/glue/thread_wrapper.h" |
| #include "media/base/media_permission.h" |
| #include "media/media_buildflags.h" |
| #include "media/video/gpu_video_accelerator_factories.h" |
| #include "third_party/blink/public/platform/web_media_constraints.h" |
| #include "third_party/blink/public/platform/web_media_stream.h" |
| #include "third_party/blink/public/platform/web_media_stream_source.h" |
| #include "third_party/blink/public/platform/web_media_stream_track.h" |
| #include "third_party/blink/public/platform/web_url.h" |
| #include "third_party/blink/public/web/web_document.h" |
| #include "third_party/blink/public/web/web_local_frame.h" |
| #include "third_party/webrtc/api/create_peerconnection_factory.h" |
| #include "third_party/webrtc/api/mediaconstraintsinterface.h" |
| #include "third_party/webrtc/api/videosourceproxy.h" |
| #include "third_party/webrtc/media/engine/multiplexcodecfactory.h" |
| #include "third_party/webrtc/modules/video_coding/codecs/h264/include/h264.h" |
| #include "third_party/webrtc/rtc_base/refcountedobject.h" |
| #include "third_party/webrtc/rtc_base/ssladapter.h" |
| |
| namespace content { |
| |
| namespace { |
| |
| enum WebRTCIPHandlingPolicy { |
| DEFAULT, |
| DEFAULT_PUBLIC_AND_PRIVATE_INTERFACES, |
| DEFAULT_PUBLIC_INTERFACE_ONLY, |
| DISABLE_NON_PROXIED_UDP, |
| }; |
| |
| WebRTCIPHandlingPolicy GetWebRTCIPHandlingPolicy( |
| const std::string& preference) { |
| if (preference == kWebRTCIPHandlingDefaultPublicAndPrivateInterfaces) |
| return DEFAULT_PUBLIC_AND_PRIVATE_INTERFACES; |
| if (preference == kWebRTCIPHandlingDefaultPublicInterfaceOnly) |
| return DEFAULT_PUBLIC_INTERFACE_ONLY; |
| if (preference == kWebRTCIPHandlingDisableNonProxiedUdp) |
| return DISABLE_NON_PROXIED_UDP; |
| return DEFAULT; |
| } |
| |
| bool IsValidPortRange(uint16_t min_port, uint16_t max_port) { |
| DCHECK(min_port <= max_port); |
| return min_port != 0 && max_port != 0; |
| } |
| |
| // PeerConnectionDependencies wants to own the factory, so we provide a simple |
| // object that delegates calls to the IpcPacketSocketFactory. |
| // TODO(zstein): Move the creation logic from IpcPacketSocketFactory in to this |
| // class. |
| class ProxyAsyncResolverFactory final : public webrtc::AsyncResolverFactory { |
| public: |
| ProxyAsyncResolverFactory(IpcPacketSocketFactory* ipc_psf) |
| : ipc_psf_(ipc_psf) { |
| DCHECK(ipc_psf); |
| } |
| |
| rtc::AsyncResolverInterface* Create() override { |
| return ipc_psf_->CreateAsyncResolver(); |
| } |
| |
| private: |
| IpcPacketSocketFactory* ipc_psf_; |
| }; |
| |
| } // namespace |
| |
| PeerConnectionDependencyFactory::PeerConnectionDependencyFactory( |
| P2PSocketDispatcher* p2p_socket_dispatcher) |
| : network_manager_(nullptr), |
| p2p_socket_dispatcher_(p2p_socket_dispatcher), |
| signaling_thread_(nullptr), |
| worker_thread_(nullptr), |
| chrome_signaling_thread_("Chrome_libJingle_Signaling"), |
| chrome_worker_thread_("Chrome_libJingle_WorkerThread") { |
| TryScheduleStunProbeTrial(); |
| } |
| |
| PeerConnectionDependencyFactory::~PeerConnectionDependencyFactory() { |
| DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_); |
| DVLOG(1) << "~PeerConnectionDependencyFactory()"; |
| DCHECK(!pc_factory_); |
| } |
| |
| std::unique_ptr<blink::WebRTCPeerConnectionHandler> |
| PeerConnectionDependencyFactory::CreateRTCPeerConnectionHandler( |
| blink::WebRTCPeerConnectionHandlerClient* client, |
| scoped_refptr<base::SingleThreadTaskRunner> task_runner) { |
| // 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(blink::WebRTCAPIName::kRTCPeerConnection); |
| |
| return std::make_unique<RTCPeerConnectionHandler>(client, this, task_runner); |
| } |
| |
| 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()"; |
| |
| #if BUILDFLAG(RTC_USE_H264) && BUILDFLAG(ENABLE_FFMPEG_VIDEO_DECODERS) |
| // Building /w |rtc_use_h264|, is the corresponding run-time feature enabled? |
| if (!base::FeatureList::IsEnabled(kWebRtcH264WithOpenH264FFmpeg)) { |
| // Feature is to be disabled. |
| webrtc::DisableRtcUseH264(); |
| } |
| #else |
| webrtc::DisableRtcUseH264(); |
| #endif // BUILDFLAG(RTC_USE_H264) && BUILDFLAG(ENABLE_FFMPEG_VIDEO_DECODERS) |
| |
| base::MessageLoopCurrent::Get()->AddDestructionObserver(this); |
| // To allow sending to the signaling/worker threads. |
| jingle_glue::JingleThreadWrapper::EnsureForCurrentMessageLoop(); |
| jingle_glue::JingleThreadWrapper::current()->set_send_allowed(true); |
| |
| EnsureWebRtcAudioDeviceImpl(); |
| |
| CHECK(chrome_signaling_thread_.Start()); |
| CHECK(chrome_worker_thread_.Start()); |
| |
| base::WaitableEvent start_worker_event( |
| base::WaitableEvent::ResetPolicy::MANUAL, |
| base::WaitableEvent::InitialState::NOT_SIGNALED); |
| chrome_worker_thread_.task_runner()->PostTask( |
| FROM_HERE, |
| base::BindOnce(&PeerConnectionDependencyFactory::InitializeWorkerThread, |
| base::Unretained(this), &worker_thread_, |
| &start_worker_event)); |
| |
| base::WaitableEvent create_network_manager_event( |
| base::WaitableEvent::ResetPolicy::MANUAL, |
| base::WaitableEvent::InitialState::NOT_SIGNALED); |
| std::unique_ptr<MdnsResponderAdapter> mdns_responder; |
| #if BUILDFLAG(ENABLE_MDNS) |
| if (base::FeatureList::IsEnabled(features::kWebRtcHideLocalIpsWithMdns)) { |
| mdns_responder = std::make_unique<MdnsResponderAdapter>(); |
| } |
| #endif // BUILDFLAG(ENABLE_MDNS) |
| chrome_worker_thread_.task_runner()->PostTask( |
| FROM_HERE, |
| base::BindOnce(&PeerConnectionDependencyFactory:: |
| CreateIpcNetworkManagerOnWorkerThread, |
| base::Unretained(this), &create_network_manager_event, |
| std::move(mdns_responder))); |
| |
| start_worker_event.Wait(); |
| create_network_manager_event.Wait(); |
| |
| CHECK(worker_thread_); |
| |
| // Init SSL, which will be needed by PeerConnection. |
| if (!rtc::InitializeSSL()) { |
| LOG(ERROR) << "Failed on InitializeSSL."; |
| NOTREACHED(); |
| return; |
| } |
| |
| base::WaitableEvent start_signaling_event( |
| base::WaitableEvent::ResetPolicy::MANUAL, |
| base::WaitableEvent::InitialState::NOT_SIGNALED); |
| chrome_signaling_thread_.task_runner()->PostTask( |
| FROM_HERE, |
| base::BindOnce( |
| &PeerConnectionDependencyFactory::InitializeSignalingThread, |
| base::Unretained(this), |
| RenderThreadImpl::current()->GetGpuFactories(), |
| &start_signaling_event)); |
| |
| start_signaling_event.Wait(); |
| CHECK(signaling_thread_); |
| } |
| |
| void PeerConnectionDependencyFactory::InitializeSignalingThread( |
| 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(); |
| |
| net::NetworkTrafficAnnotationTag traffic_annotation = |
| net::DefineNetworkTrafficAnnotation("webrtc_peer_connection", R"( |
| semantics { |
| sender: "WebRTC" |
| description: |
| "WebRTC is an API that provides web applications with Real Time " |
| "Communication (RTC) capabilities. It is used to establish a " |
| "secure session with a remote peer, transmitting and receiving " |
| "audio, video and potentially other data." |
| trigger: |
| "Application creates an RTCPeerConnection and connects it to a " |
| "remote peer by exchanging an SDP offer and answer." |
| data: |
| "Media encrypted using DTLS-SRTP, and protocol-level messages for " |
| "the various subprotocols employed by WebRTC (including ICE, DTLS, " |
| "RTCP, etc.). Note that ICE connectivity checks may leak the " |
| "user's IP address(es), subject to the restrictions/guidance in " |
| "https://datatracker.ietf.org/doc/draft-ietf-rtcweb-ip-handling." |
| destination: OTHER |
| destination_other: |
| "A destination determined by the web application that created the " |
| "connection." |
| } |
| policy { |
| cookies_allowed: NO |
| setting: |
| "This feature cannot be disabled in settings, but it won't be used " |
| "unless the application creates an RTCPeerConnection. Media can " |
| "only be captured with user's consent, but data may be sent " |
| "withouth that." |
| policy_exception_justification: |
| "Not implemented. 'WebRtcUdpPortRange' policy can limit the range " |
| "of ports used by WebRTC, but there is no policy to generally " |
| "block it." |
| } |
| )"); |
| socket_factory_.reset(new IpcPacketSocketFactory(p2p_socket_dispatcher_.get(), |
| traffic_annotation)); |
| |
| const base::CommandLine* cmd_line = base::CommandLine::ForCurrentProcess(); |
| |
| std::unique_ptr<webrtc::VideoEncoderFactory> webrtc_encoder_factory = |
| CreateWebrtcVideoEncoderFactory(gpu_factories); |
| std::unique_ptr<webrtc::VideoDecoderFactory> webrtc_decoder_factory = |
| CreateWebrtcVideoDecoderFactory(gpu_factories); |
| |
| // Enable Multiplex codec in SDP optionally. |
| if (base::FeatureList::IsEnabled(features::kWebRtcMultiplexCodec)) { |
| webrtc_encoder_factory = std::make_unique<webrtc::MultiplexEncoderFactory>( |
| std::move(webrtc_encoder_factory)); |
| webrtc_decoder_factory = std::make_unique<webrtc::MultiplexDecoderFactory>( |
| std::move(webrtc_decoder_factory)); |
| } |
| |
| pc_factory_ = webrtc::CreatePeerConnectionFactory( |
| worker_thread_ /* network thread */, worker_thread_, signaling_thread_, |
| audio_device_.get(), CreateWebrtcAudioEncoderFactory(), |
| CreateWebrtcAudioDecoderFactory(), std::move(webrtc_encoder_factory), |
| std::move(webrtc_decoder_factory), nullptr /* audio_mixer */, |
| nullptr /* audio_processing */); |
| 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() != nullptr; |
| } |
| |
| scoped_refptr<webrtc::PeerConnectionInterface> |
| PeerConnectionDependencyFactory::CreatePeerConnection( |
| const webrtc::PeerConnectionInterface::RTCConfiguration& config, |
| blink::WebLocalFrame* web_frame, |
| webrtc::PeerConnectionObserver* observer) { |
| CHECK(web_frame); |
| CHECK(observer); |
| if (!GetPcFactory().get()) |
| return nullptr; |
| |
| webrtc::PeerConnectionDependencies dependencies(observer); |
| dependencies.allocator = CreatePortAllocator(web_frame); |
| dependencies.async_resolver_factory = |
| std::make_unique<ProxyAsyncResolverFactory>(socket_factory_.get()); |
| return GetPcFactory() |
| ->CreatePeerConnection(config, std::move(dependencies)) |
| .get(); |
| } |
| |
| std::unique_ptr<P2PPortAllocator> |
| PeerConnectionDependencyFactory::CreatePortAllocator( |
| blink::WebLocalFrame* web_frame) { |
| DCHECK(web_frame); |
| |
| // Copy the flag from Preference associated with this WebLocalFrame. |
| P2PPortAllocator::Config port_config; |
| uint16_t min_port = 0; |
| uint16_t max_port = 0; |
| |
| // |media_permission| will be called to check mic/camera permission. If at |
| // least one of them is granted, P2PPortAllocator is allowed to gather local |
| // host IP addresses as ICE candidates. |media_permission| could be nullptr, |
| // which means the permission will be granted automatically. This could be the |
| // case when either the experiment is not enabled or the preference is not |
| // enforced. |
| // |
| // Note on |media_permission| lifetime: |media_permission| is owned by a frame |
| // (RenderFrameImpl). It is also stored as an indirect member of |
| // RTCPeerConnectionHandler (through PeerConnection/PeerConnectionInterface -> |
| // P2PPortAllocator -> FilteringNetworkManager -> |media_permission|). |
| // The RTCPeerConnectionHandler is owned as RTCPeerConnection::m_peerHandler |
| // in Blink, which will be reset in RTCPeerConnection::stop(). Since |
| // ActiveDOMObject::stop() is guaranteed to be called before a frame is |
| // detached, it is impossible for RTCPeerConnectionHandler to outlive the |
| // frame. Therefore using a raw pointer of |media_permission| is safe here. |
| media::MediaPermission* media_permission = nullptr; |
| if (!GetContentClient() |
| ->renderer() |
| ->ShouldEnforceWebRTCRoutingPreferences()) { |
| port_config.enable_multiple_routes = true; |
| port_config.enable_nonproxied_udp = true; |
| VLOG(3) << "WebRTC routing preferences will not be enforced"; |
| } else { |
| if (web_frame && web_frame->View()) { |
| RenderViewImpl* renderer_view_impl = |
| RenderViewImpl::FromWebView(web_frame->View()); |
| if (renderer_view_impl) { |
| // TODO(guoweis): |enable_multiple_routes| should be renamed to |
| // |request_multiple_routes|. Whether local IP addresses could be |
| // collected depends on if mic/camera permission is granted for this |
| // origin. |
| WebRTCIPHandlingPolicy policy = |
| GetWebRTCIPHandlingPolicy(renderer_view_impl->renderer_preferences() |
| .webrtc_ip_handling_policy); |
| switch (policy) { |
| // TODO(guoweis): specify the flag of disabling local candidate |
| // collection when webrtc is updated. |
| case DEFAULT_PUBLIC_INTERFACE_ONLY: |
| case DEFAULT_PUBLIC_AND_PRIVATE_INTERFACES: |
| port_config.enable_multiple_routes = false; |
| port_config.enable_nonproxied_udp = true; |
| port_config.enable_default_local_candidate = |
| (policy == DEFAULT_PUBLIC_AND_PRIVATE_INTERFACES); |
| break; |
| case DISABLE_NON_PROXIED_UDP: |
| port_config.enable_multiple_routes = false; |
| port_config.enable_nonproxied_udp = false; |
| break; |
| case DEFAULT: |
| port_config.enable_multiple_routes = true; |
| port_config.enable_nonproxied_udp = true; |
| break; |
| } |
| |
| min_port = |
| renderer_view_impl->renderer_preferences().webrtc_udp_min_port; |
| max_port = |
| renderer_view_impl->renderer_preferences().webrtc_udp_max_port; |
| |
| VLOG(3) << "WebRTC routing preferences: " |
| << "policy: " << policy |
| << ", multiple_routes: " << port_config.enable_multiple_routes |
| << ", nonproxied_udp: " << port_config.enable_nonproxied_udp |
| << ", min_udp_port: " << min_port |
| << ", max_udp_port: " << max_port; |
| } |
| } |
| if (port_config.enable_multiple_routes) { |
| bool create_media_permission = |
| base::CommandLine::ForCurrentProcess()->HasSwitch( |
| switches::kEnforceWebRtcIPPermissionCheck); |
| create_media_permission = |
| create_media_permission || |
| !StartsWith(base::FieldTrialList::FindFullName( |
| "WebRTC-LocalIPPermissionCheck"), |
| "Disabled", base::CompareCase::SENSITIVE); |
| if (create_media_permission) { |
| content::RenderFrameImpl* render_frame = |
| content::RenderFrameImpl::FromWebFrame(web_frame); |
| if (render_frame) |
| media_permission = render_frame->GetMediaPermission(); |
| DCHECK(media_permission); |
| } |
| } |
| } |
| |
| const GURL& requesting_origin = |
| GURL(web_frame->GetDocument().Url()).GetOrigin(); |
| |
| std::unique_ptr<rtc::NetworkManager> network_manager; |
| if (port_config.enable_multiple_routes) { |
| FilteringNetworkManager* filtering_network_manager = |
| new FilteringNetworkManager(network_manager_.get(), requesting_origin, |
| media_permission); |
| network_manager.reset(filtering_network_manager); |
| } else { |
| network_manager.reset(new EmptyNetworkManager(network_manager_.get())); |
| } |
| auto port_allocator = std::make_unique<P2PPortAllocator>( |
| p2p_socket_dispatcher_, std::move(network_manager), socket_factory_.get(), |
| port_config, requesting_origin); |
| if (IsValidPortRange(min_port, max_port)) |
| port_allocator->SetPortRange(min_port, max_port); |
| |
| return port_allocator; |
| } |
| |
| scoped_refptr<webrtc::MediaStreamInterface> |
| PeerConnectionDependencyFactory::CreateLocalMediaStream( |
| const std::string& label) { |
| return GetPcFactory()->CreateLocalMediaStream(label).get(); |
| } |
| |
| scoped_refptr<webrtc::VideoTrackSourceInterface> |
| PeerConnectionDependencyFactory::CreateVideoTrackSourceProxy( |
| webrtc::VideoTrackSourceInterface* source) { |
| // PeerConnectionFactory needs to be instantiated to make sure that |
| // signaling_thread_ and worker_thread_ exist. |
| if (!PeerConnectionFactoryCreated()) |
| CreatePeerConnectionFactory(); |
| |
| return webrtc::VideoTrackSourceProxy::Create(signaling_thread_, |
| worker_thread_, source) |
| .get(); |
| } |
| |
| scoped_refptr<webrtc::VideoTrackInterface> |
| PeerConnectionDependencyFactory::CreateLocalVideoTrack( |
| const std::string& id, |
| webrtc::VideoTrackSourceInterface* source) { |
| return GetPcFactory()->CreateVideoTrack(id, source).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, nullptr); |
| } |
| |
| WebRtcAudioDeviceImpl* |
| PeerConnectionDependencyFactory::GetWebRtcAudioDevice() { |
| DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_); |
| EnsureWebRtcAudioDeviceImpl(); |
| 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::TryScheduleStunProbeTrial() { |
| const base::CommandLine* cmd_line = base::CommandLine::ForCurrentProcess(); |
| |
| if (!cmd_line->HasSwitch(switches::kWebRtcStunProbeTrialParameter)) |
| return; |
| |
| GetPcFactory(); |
| |
| const std::string params = |
| cmd_line->GetSwitchValueASCII(switches::kWebRtcStunProbeTrialParameter); |
| |
| chrome_worker_thread_.task_runner()->PostDelayedTask( |
| FROM_HERE, |
| base::BindOnce( |
| &PeerConnectionDependencyFactory::StartStunProbeTrialOnWorkerThread, |
| base::Unretained(this), params), |
| base::TimeDelta::FromMilliseconds(kExperimentStartDelayMs)); |
| } |
| |
| void PeerConnectionDependencyFactory::StartStunProbeTrialOnWorkerThread( |
| const std::string& params) { |
| DCHECK(network_manager_); |
| DCHECK(chrome_worker_thread_.task_runner()->BelongsToCurrentThread()); |
| stun_trial_.reset(new StunProberTrial(network_manager_.get(), params, |
| socket_factory_.get())); |
| } |
| |
| void PeerConnectionDependencyFactory::CreateIpcNetworkManagerOnWorkerThread( |
| base::WaitableEvent* event, |
| std::unique_ptr<MdnsResponderAdapter> mdns_responder) { |
| DCHECK(chrome_worker_thread_.task_runner()->BelongsToCurrentThread()); |
| network_manager_ = std::make_unique<IpcNetworkManager>( |
| p2p_socket_dispatcher_.get(), std::move(mdns_responder)); |
| event->Signal(); |
| } |
| |
| void PeerConnectionDependencyFactory::DeleteIpcNetworkManager() { |
| DCHECK(chrome_worker_thread_.task_runner()->BelongsToCurrentThread()); |
| network_manager_.reset(); |
| } |
| |
| void PeerConnectionDependencyFactory::CleanupPeerConnectionFactory() { |
| DVLOG(1) << "PeerConnectionDependencyFactory::CleanupPeerConnectionFactory()"; |
| pc_factory_ = nullptr; |
| 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_.task_runner()->PostTask( |
| FROM_HERE, |
| base::BindOnce( |
| &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."; |
| } |
| } |
| } |
| |
| void PeerConnectionDependencyFactory::EnsureInitialized() { |
| DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_); |
| GetPcFactory(); |
| } |
| |
| scoped_refptr<base::SingleThreadTaskRunner> |
| PeerConnectionDependencyFactory::GetWebRtcWorkerThread() const { |
| DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_); |
| return chrome_worker_thread_.IsRunning() ? chrome_worker_thread_.task_runner() |
| : nullptr; |
| } |
| |
| rtc::Thread* PeerConnectionDependencyFactory::GetWebRtcWorkerThreadRtcThread() |
| const { |
| DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_); |
| return chrome_worker_thread_.IsRunning() ? worker_thread_ : nullptr; |
| } |
| |
| scoped_refptr<base::SingleThreadTaskRunner> |
| PeerConnectionDependencyFactory::GetWebRtcSignalingThread() const { |
| DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_); |
| return chrome_signaling_thread_.IsRunning() |
| ? chrome_signaling_thread_.task_runner() |
| : nullptr; |
| } |
| |
| void PeerConnectionDependencyFactory::EnsureWebRtcAudioDeviceImpl() { |
| DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_); |
| if (audio_device_.get()) |
| return; |
| |
| audio_device_ = new rtc::RefCountedObject<WebRtcAudioDeviceImpl>(); |
| } |
| |
| std::unique_ptr<webrtc::RtpCapabilities> |
| PeerConnectionDependencyFactory::GetSenderCapabilities( |
| const std::string& kind) { |
| if (kind == "audio") { |
| return std::make_unique<webrtc::RtpCapabilities>( |
| GetPcFactory()->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_AUDIO)); |
| } else if (kind == "video") { |
| return std::make_unique<webrtc::RtpCapabilities>( |
| GetPcFactory()->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO)); |
| } |
| return nullptr; |
| } |
| |
| std::unique_ptr<webrtc::RtpCapabilities> |
| PeerConnectionDependencyFactory::GetReceiverCapabilities( |
| const std::string& kind) { |
| if (kind == "audio") { |
| return std::make_unique<webrtc::RtpCapabilities>( |
| GetPcFactory()->GetRtpReceiverCapabilities(cricket::MEDIA_TYPE_AUDIO)); |
| } else if (kind == "video") { |
| return std::make_unique<webrtc::RtpCapabilities>( |
| GetPcFactory()->GetRtpReceiverCapabilities(cricket::MEDIA_TYPE_VIDEO)); |
| } |
| return nullptr; |
| } |
| |
| } // namespace content |