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710 lines
29 KiB
C++
710 lines
29 KiB
C++
/*
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* Copyright 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef PC_PEER_CONNECTION_H_
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#define PC_PEER_CONNECTION_H_
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#include <stdint.h>
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#include <functional>
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#include <map>
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#include <memory>
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#include <set>
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#include <string>
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#include <utility>
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#include <vector>
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#include "absl/types/optional.h"
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#include "api/adaptation/resource.h"
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#include "api/async_dns_resolver.h"
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#include "api/async_resolver_factory.h"
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#include "api/audio_options.h"
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#include "api/candidate.h"
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#include "api/crypto/crypto_options.h"
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#include "api/data_channel_interface.h"
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#include "api/dtls_transport_interface.h"
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#include "api/ice_transport_interface.h"
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#include "api/jsep.h"
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#include "api/media_stream_interface.h"
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#include "api/media_types.h"
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#include "api/packet_socket_factory.h"
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#include "api/peer_connection_interface.h"
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#include "api/rtc_error.h"
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#include "api/rtc_event_log/rtc_event_log.h"
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#include "api/rtc_event_log_output.h"
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#include "api/rtp_parameters.h"
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#include "api/rtp_receiver_interface.h"
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#include "api/rtp_sender_interface.h"
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#include "api/rtp_transceiver_interface.h"
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#include "api/scoped_refptr.h"
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#include "api/sctp_transport_interface.h"
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#include "api/sequence_checker.h"
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#include "api/set_local_description_observer_interface.h"
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#include "api/set_remote_description_observer_interface.h"
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#include "api/stats/rtc_stats_collector_callback.h"
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#include "api/transport/bitrate_settings.h"
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#include "api/transport/data_channel_transport_interface.h"
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#include "api/transport/enums.h"
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#include "api/turn_customizer.h"
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#include "api/video/video_bitrate_allocator_factory.h"
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#include "call/call.h"
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#include "media/base/media_channel.h"
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#include "media/base/media_engine.h"
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#include "p2p/base/ice_transport_internal.h"
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#include "p2p/base/port.h"
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#include "p2p/base/port_allocator.h"
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#include "p2p/base/transport_description.h"
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#include "pc/channel.h"
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#include "pc/channel_interface.h"
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#include "pc/channel_manager.h"
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#include "pc/connection_context.h"
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#include "pc/data_channel_controller.h"
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#include "pc/data_channel_utils.h"
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#include "pc/dtls_transport.h"
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#include "pc/jsep_transport_controller.h"
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#include "pc/peer_connection_internal.h"
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#include "pc/peer_connection_message_handler.h"
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#include "pc/rtc_stats_collector.h"
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#include "pc/rtp_receiver.h"
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#include "pc/rtp_sender.h"
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#include "pc/rtp_transceiver.h"
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#include "pc/rtp_transmission_manager.h"
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#include "pc/rtp_transport_internal.h"
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#include "pc/sctp_data_channel.h"
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#include "pc/sctp_transport.h"
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#include "pc/sdp_offer_answer.h"
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#include "pc/session_description.h"
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#include "pc/stats_collector.h"
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#include "pc/stream_collection.h"
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#include "pc/transceiver_list.h"
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#include "pc/transport_stats.h"
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#include "pc/usage_pattern.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/copy_on_write_buffer.h"
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#include "rtc_base/network/sent_packet.h"
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#include "rtc_base/rtc_certificate.h"
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#include "rtc_base/ssl_certificate.h"
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#include "rtc_base/ssl_stream_adapter.h"
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#include "rtc_base/task_utils/pending_task_safety_flag.h"
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#include "rtc_base/third_party/sigslot/sigslot.h"
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#include "rtc_base/thread.h"
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#include "rtc_base/thread_annotations.h"
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#include "rtc_base/unique_id_generator.h"
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#include "rtc_base/weak_ptr.h"
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namespace webrtc {
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// PeerConnection is the implementation of the PeerConnection object as defined
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// by the PeerConnectionInterface API surface.
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// The class currently is solely responsible for the following:
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// - Managing the session state machine (signaling state).
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// - Creating and initializing lower-level objects, like PortAllocator and
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// BaseChannels.
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// - Owning and managing the life cycle of the RtpSender/RtpReceiver and track
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// objects.
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// - Tracking the current and pending local/remote session descriptions.
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// The class currently is jointly responsible for the following:
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// - Parsing and interpreting SDP.
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// - Generating offers and answers based on the current state.
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// - The ICE state machine.
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// - Generating stats.
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class PeerConnection : public PeerConnectionInternal,
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public JsepTransportController::Observer,
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public sigslot::has_slots<> {
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public:
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// Creates a PeerConnection and initializes it with the given values.
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// If the initialization fails, the function releases the PeerConnection
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// and returns nullptr.
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//
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// Note that the function takes ownership of dependencies, and will
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// either use them or release them, whether it succeeds or fails.
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static RTCErrorOr<rtc::scoped_refptr<PeerConnection>> Create(
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rtc::scoped_refptr<ConnectionContext> context,
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const PeerConnectionFactoryInterface::Options& options,
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std::unique_ptr<RtcEventLog> event_log,
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std::unique_ptr<Call> call,
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const PeerConnectionInterface::RTCConfiguration& configuration,
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PeerConnectionDependencies dependencies);
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rtc::scoped_refptr<StreamCollectionInterface> local_streams() override;
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rtc::scoped_refptr<StreamCollectionInterface> remote_streams() override;
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bool AddStream(MediaStreamInterface* local_stream) override;
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void RemoveStream(MediaStreamInterface* local_stream) override;
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RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>> AddTrack(
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rtc::scoped_refptr<MediaStreamTrackInterface> track,
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const std::vector<std::string>& stream_ids) override;
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bool RemoveTrack(RtpSenderInterface* sender) override;
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RTCError RemoveTrackNew(
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rtc::scoped_refptr<RtpSenderInterface> sender) override;
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RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> AddTransceiver(
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rtc::scoped_refptr<MediaStreamTrackInterface> track) override;
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RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> AddTransceiver(
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rtc::scoped_refptr<MediaStreamTrackInterface> track,
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const RtpTransceiverInit& init) override;
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RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> AddTransceiver(
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cricket::MediaType media_type) override;
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RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> AddTransceiver(
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cricket::MediaType media_type,
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const RtpTransceiverInit& init) override;
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rtc::scoped_refptr<RtpSenderInterface> CreateSender(
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const std::string& kind,
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const std::string& stream_id) override;
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std::vector<rtc::scoped_refptr<RtpSenderInterface>> GetSenders()
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const override;
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std::vector<rtc::scoped_refptr<RtpReceiverInterface>> GetReceivers()
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const override;
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std::vector<rtc::scoped_refptr<RtpTransceiverInterface>> GetTransceivers()
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const override;
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RTCErrorOr<rtc::scoped_refptr<DataChannelInterface>> CreateDataChannelOrError(
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const std::string& label,
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const DataChannelInit* config) override;
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// WARNING: LEGACY. See peerconnectioninterface.h
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bool GetStats(StatsObserver* observer,
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webrtc::MediaStreamTrackInterface* track,
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StatsOutputLevel level) override;
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// Spec-complaint GetStats(). See peerconnectioninterface.h
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void GetStats(RTCStatsCollectorCallback* callback) override;
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void GetStats(
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rtc::scoped_refptr<RtpSenderInterface> selector,
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rtc::scoped_refptr<RTCStatsCollectorCallback> callback) override;
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void GetStats(
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rtc::scoped_refptr<RtpReceiverInterface> selector,
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rtc::scoped_refptr<RTCStatsCollectorCallback> callback) override;
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void ClearStatsCache() override;
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SignalingState signaling_state() override;
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IceConnectionState ice_connection_state() override;
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IceConnectionState standardized_ice_connection_state() override;
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PeerConnectionState peer_connection_state() override;
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IceGatheringState ice_gathering_state() override;
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absl::optional<bool> can_trickle_ice_candidates() override;
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const SessionDescriptionInterface* local_description() const override;
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const SessionDescriptionInterface* remote_description() const override;
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const SessionDescriptionInterface* current_local_description() const override;
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const SessionDescriptionInterface* current_remote_description()
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const override;
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const SessionDescriptionInterface* pending_local_description() const override;
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const SessionDescriptionInterface* pending_remote_description()
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const override;
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void RestartIce() override;
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// JSEP01
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void CreateOffer(CreateSessionDescriptionObserver* observer,
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const RTCOfferAnswerOptions& options) override;
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void CreateAnswer(CreateSessionDescriptionObserver* observer,
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const RTCOfferAnswerOptions& options) override;
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void SetLocalDescription(
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std::unique_ptr<SessionDescriptionInterface> desc,
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rtc::scoped_refptr<SetLocalDescriptionObserverInterface> observer)
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override;
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void SetLocalDescription(
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rtc::scoped_refptr<SetLocalDescriptionObserverInterface> observer)
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override;
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// TODO(https://crbug.com/webrtc/11798): Delete these methods in favor of the
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// ones taking SetLocalDescriptionObserverInterface as argument.
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void SetLocalDescription(SetSessionDescriptionObserver* observer,
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SessionDescriptionInterface* desc) override;
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void SetLocalDescription(SetSessionDescriptionObserver* observer) override;
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void SetRemoteDescription(
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std::unique_ptr<SessionDescriptionInterface> desc,
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rtc::scoped_refptr<SetRemoteDescriptionObserverInterface> observer)
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override;
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// TODO(https://crbug.com/webrtc/11798): Delete this methods in favor of the
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// ones taking SetRemoteDescriptionObserverInterface as argument.
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void SetRemoteDescription(SetSessionDescriptionObserver* observer,
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SessionDescriptionInterface* desc) override;
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PeerConnectionInterface::RTCConfiguration GetConfiguration() override;
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RTCError SetConfiguration(
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const PeerConnectionInterface::RTCConfiguration& configuration) override;
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bool AddIceCandidate(const IceCandidateInterface* candidate) override;
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void AddIceCandidate(std::unique_ptr<IceCandidateInterface> candidate,
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std::function<void(RTCError)> callback) override;
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bool RemoveIceCandidates(
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const std::vector<cricket::Candidate>& candidates) override;
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RTCError SetBitrate(const BitrateSettings& bitrate) override;
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void SetAudioPlayout(bool playout) override;
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void SetAudioRecording(bool recording) override;
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rtc::scoped_refptr<DtlsTransportInterface> LookupDtlsTransportByMid(
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const std::string& mid) override;
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rtc::scoped_refptr<DtlsTransport> LookupDtlsTransportByMidInternal(
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const std::string& mid);
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rtc::scoped_refptr<SctpTransportInterface> GetSctpTransport() const override;
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void AddAdaptationResource(rtc::scoped_refptr<Resource> resource) override;
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bool StartRtcEventLog(std::unique_ptr<RtcEventLogOutput> output,
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int64_t output_period_ms) override;
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bool StartRtcEventLog(std::unique_ptr<RtcEventLogOutput> output) override;
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void StopRtcEventLog() override;
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void Close() override;
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rtc::Thread* signaling_thread() const final {
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return context_->signaling_thread();
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}
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// PeerConnectionInternal implementation.
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rtc::Thread* network_thread() const final {
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return context_->network_thread();
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}
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rtc::Thread* worker_thread() const final { return context_->worker_thread(); }
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std::string session_id() const override {
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return session_id_;
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}
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bool initial_offerer() const override {
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RTC_DCHECK_RUN_ON(signaling_thread());
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return transport_controller_ && transport_controller_->initial_offerer();
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}
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std::vector<
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rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
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GetTransceiversInternal() const override {
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RTC_DCHECK_RUN_ON(signaling_thread());
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return rtp_manager()->transceivers()->List();
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}
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sigslot::signal1<SctpDataChannel*>& SignalSctpDataChannelCreated() override {
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RTC_DCHECK_RUN_ON(signaling_thread());
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return data_channel_controller_.SignalSctpDataChannelCreated();
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}
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std::vector<DataChannelStats> GetDataChannelStats() const override;
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absl::optional<std::string> sctp_transport_name() const override;
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absl::optional<std::string> sctp_mid() const override;
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cricket::CandidateStatsList GetPooledCandidateStats() const override;
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std::map<std::string, cricket::TransportStats> GetTransportStatsByNames(
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const std::set<std::string>& transport_names) override;
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Call::Stats GetCallStats() override;
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bool GetLocalCertificate(
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const std::string& transport_name,
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rtc::scoped_refptr<rtc::RTCCertificate>* certificate) override;
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std::unique_ptr<rtc::SSLCertChain> GetRemoteSSLCertChain(
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const std::string& transport_name) override;
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bool IceRestartPending(const std::string& content_name) const override;
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bool NeedsIceRestart(const std::string& content_name) const override;
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bool GetSslRole(const std::string& content_name, rtc::SSLRole* role) override;
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// Functions needed by DataChannelController
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void NoteDataAddedEvent() { NoteUsageEvent(UsageEvent::DATA_ADDED); }
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// Returns the observer. Will crash on CHECK if the observer is removed.
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PeerConnectionObserver* Observer() const;
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bool IsClosed() const {
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RTC_DCHECK_RUN_ON(signaling_thread());
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return !sdp_handler_ ||
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sdp_handler_->signaling_state() == PeerConnectionInterface::kClosed;
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}
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// Get current SSL role used by SCTP's underlying transport.
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bool GetSctpSslRole(rtc::SSLRole* role);
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// Handler for the "channel closed" signal
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void OnSctpDataChannelClosed(DataChannelInterface* channel);
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bool ShouldFireNegotiationNeededEvent(uint32_t event_id) override;
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// Functions needed by SdpOfferAnswerHandler
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StatsCollector* stats() {
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RTC_DCHECK_RUN_ON(signaling_thread());
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return stats_.get();
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}
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DataChannelController* data_channel_controller() {
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RTC_DCHECK_RUN_ON(signaling_thread());
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return &data_channel_controller_;
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}
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bool dtls_enabled() const {
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RTC_DCHECK_RUN_ON(signaling_thread());
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return dtls_enabled_;
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}
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const PeerConnectionInterface::RTCConfiguration* configuration() const {
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RTC_DCHECK_RUN_ON(signaling_thread());
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return &configuration_;
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}
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PeerConnectionMessageHandler* message_handler() {
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RTC_DCHECK_RUN_ON(signaling_thread());
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return &message_handler_;
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}
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RtpTransmissionManager* rtp_manager() { return rtp_manager_.get(); }
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const RtpTransmissionManager* rtp_manager() const {
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return rtp_manager_.get();
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}
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cricket::ChannelManager* channel_manager() const;
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JsepTransportController* transport_controller() {
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return transport_controller_.get();
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}
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cricket::PortAllocator* port_allocator() { return port_allocator_.get(); }
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Call* call_ptr() { return call_ptr_; }
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ConnectionContext* context() { return context_.get(); }
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const PeerConnectionFactoryInterface::Options* options() const {
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return &options_;
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}
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void SetIceConnectionState(IceConnectionState new_state);
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void NoteUsageEvent(UsageEvent event);
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// Asynchronously adds a remote candidate on the network thread.
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void AddRemoteCandidate(const std::string& mid,
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const cricket::Candidate& candidate);
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// Report the UMA metric SdpFormatReceived for the given remote description.
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void ReportSdpFormatReceived(
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const SessionDescriptionInterface& remote_description);
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// Report the UMA metric BundleUsage for the given remote description.
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void ReportSdpBundleUsage(
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const SessionDescriptionInterface& remote_description);
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// Returns true if the PeerConnection is configured to use Unified Plan
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// semantics for creating offers/answers and setting local/remote
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// descriptions. If this is true the RtpTransceiver API will also be available
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// to the user. If this is false, Plan B semantics are assumed.
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// TODO(bugs.webrtc.org/8530): Flip the default to be Unified Plan once
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// sufficient time has passed.
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bool IsUnifiedPlan() const {
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RTC_DCHECK_RUN_ON(signaling_thread());
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return is_unified_plan_;
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}
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bool ValidateBundleSettings(
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const cricket::SessionDescription* desc,
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const std::map<std::string, const cricket::ContentGroup*>&
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bundle_groups_by_mid);
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// Returns the MID for the data section associated with the
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// SCTP data channel, if it has been set. If no data
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// channels are configured this will return nullopt.
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absl::optional<std::string> GetDataMid() const;
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void SetSctpDataMid(const std::string& mid);
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void ResetSctpDataMid();
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// Asynchronously calls SctpTransport::Start() on the network thread for
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// `sctp_mid()` if set. Called as part of setting the local description.
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void StartSctpTransport(int local_port,
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int remote_port,
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int max_message_size);
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// Returns the CryptoOptions for this PeerConnection. This will always
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// return the RTCConfiguration.crypto_options if set and will only default
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// back to the PeerConnectionFactory settings if nothing was set.
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CryptoOptions GetCryptoOptions();
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// Internal implementation for AddTransceiver family of methods. If
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// `fire_callback` is set, fires OnRenegotiationNeeded callback if successful.
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RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> AddTransceiver(
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cricket::MediaType media_type,
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rtc::scoped_refptr<MediaStreamTrackInterface> track,
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const RtpTransceiverInit& init,
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bool fire_callback = true);
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// Returns rtp transport, result can not be nullptr.
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RtpTransportInternal* GetRtpTransport(const std::string& mid);
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// Returns true if SRTP (either using DTLS-SRTP or SDES) is required by
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// this session.
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bool SrtpRequired() const;
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bool SetupDataChannelTransport_n(const std::string& mid)
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RTC_RUN_ON(network_thread());
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void TeardownDataChannelTransport_n() RTC_RUN_ON(network_thread());
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cricket::ChannelInterface* GetChannel(const std::string& content_name)
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RTC_RUN_ON(network_thread());
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// Functions made public for testing.
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void ReturnHistogramVeryQuicklyForTesting() {
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RTC_DCHECK_RUN_ON(signaling_thread());
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return_histogram_very_quickly_ = true;
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}
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void RequestUsagePatternReportForTesting();
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protected:
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// Available for rtc::scoped_refptr creation
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PeerConnection(rtc::scoped_refptr<ConnectionContext> context,
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const PeerConnectionFactoryInterface::Options& options,
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bool is_unified_plan,
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std::unique_ptr<RtcEventLog> event_log,
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std::unique_ptr<Call> call,
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PeerConnectionDependencies& dependencies,
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bool dtls_enabled);
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~PeerConnection() override;
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private:
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RTCError Initialize(
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const PeerConnectionInterface::RTCConfiguration& configuration,
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PeerConnectionDependencies dependencies);
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void InitializeTransportController_n(
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const RTCConfiguration& configuration,
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const PeerConnectionDependencies& dependencies)
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RTC_RUN_ON(network_thread());
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rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
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FindTransceiverBySender(rtc::scoped_refptr<RtpSenderInterface> sender)
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RTC_RUN_ON(signaling_thread());
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void SetStandardizedIceConnectionState(
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PeerConnectionInterface::IceConnectionState new_state)
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RTC_RUN_ON(signaling_thread());
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void SetConnectionState(
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PeerConnectionInterface::PeerConnectionState new_state)
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RTC_RUN_ON(signaling_thread());
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// Called any time the IceGatheringState changes.
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void OnIceGatheringChange(IceGatheringState new_state)
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RTC_RUN_ON(signaling_thread());
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// New ICE candidate has been gathered.
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void OnIceCandidate(std::unique_ptr<IceCandidateInterface> candidate)
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RTC_RUN_ON(signaling_thread());
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// Gathering of an ICE candidate failed.
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void OnIceCandidateError(const std::string& address,
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int port,
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const std::string& url,
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int error_code,
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const std::string& error_text)
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RTC_RUN_ON(signaling_thread());
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// Some local ICE candidates have been removed.
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void OnIceCandidatesRemoved(const std::vector<cricket::Candidate>& candidates)
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RTC_RUN_ON(signaling_thread());
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void OnSelectedCandidatePairChanged(
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const cricket::CandidatePairChangeEvent& event)
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RTC_RUN_ON(signaling_thread());
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void OnNegotiationNeeded();
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// Returns the specified SCTP DataChannel in sctp_data_channels_,
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// or nullptr if not found.
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SctpDataChannel* FindDataChannelBySid(int sid) const
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RTC_RUN_ON(signaling_thread());
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// Called when first configuring the port allocator.
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struct InitializePortAllocatorResult {
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bool enable_ipv6;
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};
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InitializePortAllocatorResult InitializePortAllocator_n(
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const cricket::ServerAddresses& stun_servers,
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|
const std::vector<cricket::RelayServerConfig>& turn_servers,
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|
const RTCConfiguration& configuration);
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// Called when SetConfiguration is called to apply the supported subset
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|
// of the configuration on the network thread.
|
|
bool ReconfigurePortAllocator_n(
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|
const cricket::ServerAddresses& stun_servers,
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|
const std::vector<cricket::RelayServerConfig>& turn_servers,
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|
IceTransportsType type,
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|
int candidate_pool_size,
|
|
PortPrunePolicy turn_port_prune_policy,
|
|
webrtc::TurnCustomizer* turn_customizer,
|
|
absl::optional<int> stun_candidate_keepalive_interval,
|
|
bool have_local_description);
|
|
|
|
// Starts output of an RTC event log to the given output object.
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|
// This function should only be called from the worker thread.
|
|
bool StartRtcEventLog_w(std::unique_ptr<RtcEventLogOutput> output,
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|
int64_t output_period_ms);
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|
|
|
// Stops recording an RTC event log.
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|
// This function should only be called from the worker thread.
|
|
void StopRtcEventLog_w();
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|
|
|
// Returns true and the TransportInfo of the given `content_name`
|
|
// from `description`. Returns false if it's not available.
|
|
static bool GetTransportDescription(
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|
const cricket::SessionDescription* description,
|
|
const std::string& content_name,
|
|
cricket::TransportDescription* info);
|
|
|
|
// Returns the media index for a local ice candidate given the content name.
|
|
// Returns false if the local session description does not have a media
|
|
// content called `content_name`.
|
|
bool GetLocalCandidateMediaIndex(const std::string& content_name,
|
|
int* sdp_mline_index)
|
|
RTC_RUN_ON(signaling_thread());
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|
|
|
// JsepTransportController signal handlers.
|
|
void OnTransportControllerConnectionState(cricket::IceConnectionState state)
|
|
RTC_RUN_ON(signaling_thread());
|
|
void OnTransportControllerGatheringState(cricket::IceGatheringState state)
|
|
RTC_RUN_ON(signaling_thread());
|
|
void OnTransportControllerCandidatesGathered(
|
|
const std::string& transport_name,
|
|
const std::vector<cricket::Candidate>& candidates)
|
|
RTC_RUN_ON(signaling_thread());
|
|
void OnTransportControllerCandidateError(
|
|
const cricket::IceCandidateErrorEvent& event)
|
|
RTC_RUN_ON(signaling_thread());
|
|
void OnTransportControllerCandidatesRemoved(
|
|
const std::vector<cricket::Candidate>& candidates)
|
|
RTC_RUN_ON(signaling_thread());
|
|
void OnTransportControllerCandidateChanged(
|
|
const cricket::CandidatePairChangeEvent& event)
|
|
RTC_RUN_ON(signaling_thread());
|
|
void OnTransportControllerDtlsHandshakeError(rtc::SSLHandshakeError error);
|
|
|
|
// Invoked when TransportController connection completion is signaled.
|
|
// Reports stats for all transports in use.
|
|
void ReportTransportStats() RTC_RUN_ON(network_thread());
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|
|
|
// Gather the usage of IPv4/IPv6 as best connection.
|
|
static void ReportBestConnectionState(const cricket::TransportStats& stats);
|
|
|
|
static void ReportNegotiatedCiphers(
|
|
bool dtls_enabled,
|
|
const cricket::TransportStats& stats,
|
|
const std::set<cricket::MediaType>& media_types);
|
|
void ReportIceCandidateCollected(const cricket::Candidate& candidate)
|
|
RTC_RUN_ON(signaling_thread());
|
|
|
|
void ReportUsagePattern() const RTC_RUN_ON(signaling_thread());
|
|
|
|
void ReportRemoteIceCandidateAdded(const cricket::Candidate& candidate);
|
|
|
|
// JsepTransportController::Observer override.
|
|
//
|
|
// Called by `transport_controller_` when processing transport information
|
|
// from a session description, and the mapping from m= sections to transports
|
|
// changed (as a result of BUNDLE negotiation, or m= sections being
|
|
// rejected).
|
|
bool OnTransportChanged(
|
|
const std::string& mid,
|
|
RtpTransportInternal* rtp_transport,
|
|
rtc::scoped_refptr<DtlsTransport> dtls_transport,
|
|
DataChannelTransportInterface* data_channel_transport) override;
|
|
|
|
std::function<void(const rtc::CopyOnWriteBuffer& packet,
|
|
int64_t packet_time_us)>
|
|
InitializeRtcpCallback();
|
|
|
|
const rtc::scoped_refptr<ConnectionContext> context_;
|
|
const PeerConnectionFactoryInterface::Options options_;
|
|
PeerConnectionObserver* observer_ RTC_GUARDED_BY(signaling_thread()) =
|
|
nullptr;
|
|
|
|
const bool is_unified_plan_;
|
|
|
|
// The EventLog needs to outlive `call_` (and any other object that uses it).
|
|
std::unique_ptr<RtcEventLog> event_log_ RTC_GUARDED_BY(worker_thread());
|
|
|
|
// Points to the same thing as `event_log_`. Since it's const, we may read the
|
|
// pointer (but not touch the object) from any thread.
|
|
RtcEventLog* const event_log_ptr_ RTC_PT_GUARDED_BY(worker_thread());
|
|
|
|
IceConnectionState ice_connection_state_ RTC_GUARDED_BY(signaling_thread()) =
|
|
kIceConnectionNew;
|
|
PeerConnectionInterface::IceConnectionState standardized_ice_connection_state_
|
|
RTC_GUARDED_BY(signaling_thread()) = kIceConnectionNew;
|
|
PeerConnectionInterface::PeerConnectionState connection_state_
|
|
RTC_GUARDED_BY(signaling_thread()) = PeerConnectionState::kNew;
|
|
|
|
IceGatheringState ice_gathering_state_ RTC_GUARDED_BY(signaling_thread()) =
|
|
kIceGatheringNew;
|
|
PeerConnectionInterface::RTCConfiguration configuration_
|
|
RTC_GUARDED_BY(signaling_thread());
|
|
|
|
const std::unique_ptr<AsyncDnsResolverFactoryInterface>
|
|
async_dns_resolver_factory_;
|
|
std::unique_ptr<cricket::PortAllocator>
|
|
port_allocator_; // TODO(bugs.webrtc.org/9987): Accessed on both
|
|
// signaling and network thread.
|
|
const std::unique_ptr<webrtc::IceTransportFactory>
|
|
ice_transport_factory_; // TODO(bugs.webrtc.org/9987): Accessed on the
|
|
// signaling thread but the underlying raw
|
|
// pointer is given to
|
|
// `jsep_transport_controller_` and used on the
|
|
// network thread.
|
|
const std::unique_ptr<rtc::SSLCertificateVerifier> tls_cert_verifier_
|
|
RTC_GUARDED_BY(network_thread());
|
|
|
|
// The unique_ptr belongs to the worker thread, but the Call object manages
|
|
// its own thread safety.
|
|
std::unique_ptr<Call> call_ RTC_GUARDED_BY(worker_thread());
|
|
ScopedTaskSafety signaling_thread_safety_;
|
|
rtc::scoped_refptr<PendingTaskSafetyFlag> network_thread_safety_;
|
|
rtc::scoped_refptr<PendingTaskSafetyFlag> worker_thread_safety_;
|
|
|
|
// Points to the same thing as `call_`. Since it's const, we may read the
|
|
// pointer from any thread.
|
|
// TODO(bugs.webrtc.org/11992): Remove this workaround (and potential dangling
|
|
// pointer).
|
|
Call* const call_ptr_;
|
|
|
|
std::unique_ptr<StatsCollector> stats_
|
|
RTC_GUARDED_BY(signaling_thread()); // A pointer is passed to senders_
|
|
rtc::scoped_refptr<RTCStatsCollector> stats_collector_
|
|
RTC_GUARDED_BY(signaling_thread());
|
|
|
|
const std::string session_id_;
|
|
|
|
std::unique_ptr<JsepTransportController>
|
|
transport_controller_; // TODO(bugs.webrtc.org/9987): Accessed on both
|
|
// signaling and network thread.
|
|
|
|
// `sctp_mid_` is the content name (MID) in SDP.
|
|
// Note: this is used as the data channel MID by both SCTP and data channel
|
|
// transports. It is set when either transport is initialized and unset when
|
|
// both transports are deleted.
|
|
// There is one copy on the signaling thread and another copy on the
|
|
// networking thread. Changes are always initiated from the signaling
|
|
// thread, but applied first on the networking thread via an invoke().
|
|
absl::optional<std::string> sctp_mid_s_ RTC_GUARDED_BY(signaling_thread());
|
|
absl::optional<std::string> sctp_mid_n_ RTC_GUARDED_BY(network_thread());
|
|
std::string sctp_transport_name_s_ RTC_GUARDED_BY(signaling_thread());
|
|
|
|
// The machinery for handling offers and answers. Const after initialization.
|
|
std::unique_ptr<SdpOfferAnswerHandler> sdp_handler_
|
|
RTC_GUARDED_BY(signaling_thread()) RTC_PT_GUARDED_BY(signaling_thread());
|
|
|
|
const bool dtls_enabled_;
|
|
|
|
UsagePattern usage_pattern_ RTC_GUARDED_BY(signaling_thread());
|
|
bool return_histogram_very_quickly_ RTC_GUARDED_BY(signaling_thread()) =
|
|
false;
|
|
|
|
// The DataChannelController is accessed from both the signaling thread
|
|
// and networking thread. It is a thread-aware object.
|
|
DataChannelController data_channel_controller_;
|
|
|
|
// Machinery for handling messages posted to oneself
|
|
PeerConnectionMessageHandler message_handler_
|
|
RTC_GUARDED_BY(signaling_thread());
|
|
|
|
// Administration of senders, receivers and transceivers
|
|
// Accessed on both signaling and network thread. Const after Initialize().
|
|
std::unique_ptr<RtpTransmissionManager> rtp_manager_;
|
|
|
|
// Did the connectionState ever change to `connected`?
|
|
// Used to gather metrics only the first such state change.
|
|
bool was_ever_connected_ RTC_GUARDED_BY(signaling_thread()) = false;
|
|
|
|
// This variable needs to be the last one in the class.
|
|
rtc::WeakPtrFactory<PeerConnection> weak_factory_;
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // PC_PEER_CONNECTION_H_
|