| // 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 <utility> |
| #include <vector> |
| |
| #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 "build/build_config.h" |
| #include "content/common/media/media_stream_messages.h" |
| #include "content/public/common/content_client.h" |
| #include "content/public/common/content_switches.h" |
| #include "content/public/common/feature_h264_with_openh264_ffmpeg.h" |
| #include "content/public/common/features.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/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_store.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/media_stream_remote_audio_track.h" |
| #include "content/renderer/media/webrtc/stun_field_trial.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/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/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/filters/ffmpeg_glue.h" |
| #include "media/renderers/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/webrtc/api/mediaconstraintsinterface.h" |
| #include "third_party/webrtc/base/ssladapter.h" |
| #include "third_party/webrtc/modules/video_coding/codecs/h264/include/h264.h" |
| |
| #if defined(OS_ANDROID) |
| #include "media/base/android/media_codec_util.h" |
| #endif |
| |
| 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; |
| } |
| |
| } // namespace |
| |
| // Map of corresponding media constraints and platform effects. |
| struct { |
| const char* constraint; |
| const media::AudioParameters::PlatformEffectsMask effect; |
| } const kConstraintEffectMap[] = { |
| { webrtc::MediaConstraintsInterface::kGoogEchoCancellation, |
| 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; |
| } |
| } |
| } |
| } |
| |
| 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") { |
| TryScheduleStunProbeTrial(); |
| } |
| |
| 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_frame_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_frame_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()"; |
| |
| #if BUILDFLAG(RTC_USE_H264) |
| // Building /w |rtc_use_h264|, is the corresponding run-time feature enabled? |
| if (base::FeatureList::IsEnabled(kWebRtcH264WithOpenH264FFmpeg)) { |
| // |H264DecoderImpl| may be used which depends on FFmpeg, therefore we need |
| // to initialize FFmpeg before going further. |
| media::FFmpegGlue::InitializeFFmpeg(); |
| } else { |
| // Feature is to be disabled, no need to make sure FFmpeg is initialized. |
| webrtc::DisableRtcUseH264(); |
| } |
| #endif |
| |
| 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_.task_runner()->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_.task_runner()->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. |
| // |
| // TODO(davidben): BoringSSL must be initialized by Chromium code. If the |
| // initialization requirement is removed or when different libraries are |
| // allowed to call CRYPTO_library_init concurrently, remove this line and |
| // initialize within WebRTC. See https://crbug.com/542879. |
| crypto::EnsureOpenSSLInit(); |
| if (!rtc::InitializeSSL()) { |
| LOG(ERROR) << "Failed on InitializeSSL."; |
| NOTREACHED(); |
| return; |
| } |
| |
| base::WaitableEvent start_signaling_event(true, false); |
| chrome_signaling_thread_.task_runner()->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( |
| 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 && gpu_factories->IsGpuVideoAcceleratorEnabled()) { |
| 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::MediaCodecUtil::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); |
| |
| // DTLS 1.2 is the default now but could be changed to 1.0 by the experiment. |
| factory_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12; |
| std::string group_name = |
| base::FieldTrialList::FindFullName("WebRTC-PeerConnectionDTLS1.2"); |
| if (StartsWith(group_name, "Control", base::CompareCase::SENSITIVE)) |
| factory_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10; |
| |
| 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; |
| |
| rtc::scoped_ptr<PeerConnectionIdentityStore> identity_store( |
| new PeerConnectionIdentityStore( |
| base::ThreadTaskRunnerHandle::Get(), |
| GetWebRtcSignalingThread(), |
| GURL(web_frame->document().url()), |
| GURL(web_frame->document().firstPartyForCookies()))); |
| |
| // Copy the flag from Preference associated with this WebFrame. |
| P2PPortAllocator::Config port_config; |
| |
| // |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; |
| } |
| |
| VLOG(3) << "WebRTC routing preferences: " |
| << "policy: " << policy |
| << ", multiple_routes: " << port_config.enable_multiple_routes |
| << ", nonproxied_udp: " << port_config.enable_nonproxied_udp; |
| } |
| } |
| 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"), |
| "Enabled", 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->document().url()).GetOrigin(); |
| |
| scoped_ptr<rtc::NetworkManager> network_manager; |
| if (port_config.enable_multiple_routes) { |
| FilteringNetworkManager* filtering_network_manager = |
| new FilteringNetworkManager(network_manager_, requesting_origin, |
| media_permission); |
| if (media_permission) { |
| // Start permission check earlier to reduce any impact to call set up |
| // time. It's safe to use Unretained here since both destructor and |
| // Initialize can only be called on the worker thread. |
| chrome_worker_thread_.task_runner()->PostTask( |
| FROM_HERE, base::Bind(&FilteringNetworkManager::Initialize, |
| base::Unretained(filtering_network_manager))); |
| } |
| network_manager.reset(filtering_network_manager); |
| } else { |
| network_manager.reset(new EmptyNetworkManager(network_manager_)); |
| } |
| rtc::scoped_ptr<P2PPortAllocator> port_allocator(new P2PPortAllocator( |
| p2p_socket_dispatcher_, std::move(network_manager), socket_factory_.get(), |
| port_config, requesting_origin, chrome_worker_thread_.task_runner())); |
| |
| return GetPcFactory() |
| ->CreatePeerConnection(config, constraints, std::move(port_allocator), |
| std::move(identity_store), 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); |
| DCHECK(!source.remote()); |
| 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 { |
| NOTREACHED() << "Local track missing source extra data."; |
| 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::CreateRemoteAudioTrack( |
| const blink::WebMediaStreamTrack& track) { |
| blink::WebMediaStreamSource source = track.source(); |
| DCHECK_EQ(source.type(), blink::WebMediaStreamSource::TypeAudio); |
| DCHECK(source.remote()); |
| DCHECK(source.extraData()); |
| |
| blink::WebMediaStreamTrack writable_track = track; |
| writable_track.setExtraData( |
| new MediaStreamRemoteAudioTrack(source, track.isEnabled())); |
| } |
| |
| 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(*source)); |
| 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, nullptr); |
| } |
| |
| bool PeerConnectionDependencyFactory::StartRtcEventLog( |
| base::PlatformFile file) { |
| return GetPcFactory()->StartRtcEventLog(file); |
| } |
| |
| void PeerConnectionDependencyFactory::StopRtcEventLog() { |
| GetPcFactory()->StopRtcEventLog(); |
| } |
| |
| 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::TryScheduleStunProbeTrial() { |
| const base::CommandLine* cmd_line = base::CommandLine::ForCurrentProcess(); |
| |
| if (!cmd_line->HasSwitch(switches::kWebRtcStunProbeTrialParameter)) |
| return; |
| |
| // The underneath IPC channel has to be connected before sending any IPC |
| // message. |
| if (!p2p_socket_dispatcher_->connected()) { |
| base::MessageLoop::current()->PostDelayedTask( |
| FROM_HERE, |
| base::Bind(&PeerConnectionDependencyFactory::TryScheduleStunProbeTrial, |
| base::Unretained(this)), |
| base::TimeDelta::FromSeconds(1)); |
| return; |
| } |
| |
| // GetPcFactory could trigger an IPC message. If done before |
| // |p2p_socket_dispatcher_| is connected, that'll put the |
| // |p2p_socket_dispatcher_| in a bad state such that no other IPC message can |
| // be processed. |
| GetPcFactory(); |
| |
| const std::string params = |
| cmd_line->GetSwitchValueASCII(switches::kWebRtcStunProbeTrialParameter); |
| |
| chrome_worker_thread_.task_runner()->PostDelayedTask( |
| FROM_HERE, |
| base::Bind( |
| &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_, params, socket_factory_.get())); |
| } |
| |
| void PeerConnectionDependencyFactory::CreateIpcNetworkManagerOnWorkerThread( |
| base::WaitableEvent* event) { |
| DCHECK(chrome_worker_thread_.task_runner()->BelongsToCurrentThread()); |
| network_manager_ = new IpcNetworkManager(p2p_socket_dispatcher_.get()); |
| event->Signal(); |
| } |
| |
| void PeerConnectionDependencyFactory::DeleteIpcNetworkManager() { |
| DCHECK(chrome_worker_thread_.task_runner()->BelongsToCurrentThread()); |
| 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_.task_runner()->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_frame_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_frame_id, 0); |
| |
| EnsureWebRtcAudioDeviceImpl(); |
| DCHECK(GetWebRtcAudioDevice()); |
| return WebRtcAudioCapturer::CreateCapturer( |
| render_frame_id, device_info, constraints, GetWebRtcAudioDevice(), |
| audio_source); |
| } |
| |
| void PeerConnectionDependencyFactory::EnsureInitialized() { |
| DCHECK(CalledOnValidThread()); |
| GetPcFactory(); |
| } |
| |
| scoped_refptr<base::SingleThreadTaskRunner> |
| PeerConnectionDependencyFactory::GetWebRtcWorkerThread() const { |
| DCHECK(CalledOnValidThread()); |
| return chrome_worker_thread_.IsRunning() ? chrome_worker_thread_.task_runner() |
| : nullptr; |
| } |
| |
| scoped_refptr<base::SingleThreadTaskRunner> |
| PeerConnectionDependencyFactory::GetWebRtcSignalingThread() const { |
| DCHECK(CalledOnValidThread()); |
| return chrome_signaling_thread_.IsRunning() |
| ? chrome_signaling_thread_.task_runner() |
| : nullptr; |
| } |
| |
| void PeerConnectionDependencyFactory::EnsureWebRtcAudioDeviceImpl() { |
| if (audio_device_.get()) |
| return; |
| |
| audio_device_ = new WebRtcAudioDeviceImpl(); |
| } |
| |
| } // namespace content |