1344 lines
60 KiB
C++
1344 lines
60 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 <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 "api/peer_connection_interface.h"
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#include "api/transport/data_channel_transport_interface.h"
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#include "api/turn_customizer.h"
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#include "pc/data_channel_controller.h"
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#include "pc/ice_server_parsing.h"
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#include "pc/jsep_transport_controller.h"
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#include "pc/peer_connection_factory.h"
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#include "pc/peer_connection_internal.h"
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#include "pc/rtc_stats_collector.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/sctp_transport.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/webrtc_session_description_factory.h"
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#include "rtc_base/experiments/field_trial_parser.h"
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#include "rtc_base/operations_chain.h"
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#include "rtc_base/race_checker.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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class MediaStreamObserver;
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class VideoRtpReceiver;
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class RtcEventLog;
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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 RtpSenderBase::SetStreamsObserver,
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public rtc::MessageHandler,
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public sigslot::has_slots<> {
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public:
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// A bit in the usage pattern is registered when its defining event occurs at
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// least once.
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enum class UsageEvent : int {
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TURN_SERVER_ADDED = 0x01,
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STUN_SERVER_ADDED = 0x02,
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DATA_ADDED = 0x04,
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AUDIO_ADDED = 0x08,
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VIDEO_ADDED = 0x10,
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// |SetLocalDescription| returns successfully.
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SET_LOCAL_DESCRIPTION_SUCCEEDED = 0x20,
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// |SetRemoteDescription| returns successfully.
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SET_REMOTE_DESCRIPTION_SUCCEEDED = 0x40,
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// A local candidate (with type host, server-reflexive, or relay) is
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// collected.
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CANDIDATE_COLLECTED = 0x80,
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// A remote candidate is successfully added via |AddIceCandidate|.
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ADD_ICE_CANDIDATE_SUCCEEDED = 0x100,
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ICE_STATE_CONNECTED = 0x200,
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CLOSE_CALLED = 0x400,
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// A local candidate with private IP is collected.
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PRIVATE_CANDIDATE_COLLECTED = 0x800,
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// A remote candidate with private IP is added, either via AddiceCandidate
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// or from the remote description.
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REMOTE_PRIVATE_CANDIDATE_ADDED = 0x1000,
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// A local mDNS candidate is collected.
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MDNS_CANDIDATE_COLLECTED = 0x2000,
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// A remote mDNS candidate is added, either via AddIceCandidate or from the
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// remote description.
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REMOTE_MDNS_CANDIDATE_ADDED = 0x4000,
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// A local candidate with IPv6 address is collected.
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IPV6_CANDIDATE_COLLECTED = 0x8000,
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// A remote candidate with IPv6 address is added, either via AddIceCandidate
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// or from the remote description.
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REMOTE_IPV6_CANDIDATE_ADDED = 0x10000,
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// A remote candidate (with type host, server-reflexive, or relay) is
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// successfully added, either via AddIceCandidate or from the remote
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// description.
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REMOTE_CANDIDATE_ADDED = 0x20000,
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// An explicit host-host candidate pair is selected, i.e. both the local and
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// the remote candidates have the host type. This does not include candidate
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// pairs formed with equivalent prflx remote candidates, e.g. a host-prflx
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// pair where the prflx candidate has the same base as a host candidate of
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// the remote peer.
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DIRECT_CONNECTION_SELECTED = 0x40000,
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MAX_VALUE = 0x80000,
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};
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explicit PeerConnection(PeerConnectionFactory* factory,
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std::unique_ptr<RtcEventLog> event_log,
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std::unique_ptr<Call> call);
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bool Initialize(
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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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// Gets the DTLS SSL certificate associated with the audio transport on the
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// remote side. This will become populated once the DTLS connection with the
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// peer has been completed, as indicated by the ICE connection state
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// transitioning to kIceConnectionCompleted.
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// Note that this will be removed once we implement RTCDtlsTransport which
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// has standardized method for getting this information.
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// See https://www.w3.org/TR/webrtc/#rtcdtlstransport-interface
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std::unique_ptr<rtc::SSLCertificate> GetRemoteAudioSSLCertificate();
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// Version of the above method that returns the full certificate chain.
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std::unique_ptr<rtc::SSLCertChain> GetRemoteAudioSSLCertChain();
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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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rtc::scoped_refptr<DataChannelInterface> CreateDataChannel(
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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 factory_->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 factory_->network_thread();
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}
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rtc::Thread* worker_thread() const final { return factory_->worker_thread(); }
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std::string session_id() const override {
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RTC_DCHECK_RUN_ON(signaling_thread());
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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 transceivers_;
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}
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sigslot::signal1<RtpDataChannel*>& SignalRtpDataChannelCreated() override {
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return data_channel_controller_.SignalRtpDataChannelCreated();
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}
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sigslot::signal1<SctpDataChannel*>& SignalSctpDataChannelCreated() override {
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return data_channel_controller_.SignalSctpDataChannelCreated();
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}
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cricket::RtpDataChannel* rtp_data_channel() const override {
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return data_channel_controller_.rtp_data_channel();
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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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cricket::CandidateStatsList GetPooledCandidateStats() const override;
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std::map<std::string, std::string> GetTransportNamesByMid() 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 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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// 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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absl::optional<std::string> sctp_mid() {
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RTC_DCHECK_RUN_ON(signaling_thread());
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return sctp_mid_s_;
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}
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protected:
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~PeerConnection() override;
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private:
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class ImplicitCreateSessionDescriptionObserver;
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friend class ImplicitCreateSessionDescriptionObserver;
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class SetSessionDescriptionObserverAdapter;
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friend class SetSessionDescriptionObserverAdapter;
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// Represents the [[LocalIceCredentialsToReplace]] internal slot in the spec.
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// It makes the next CreateOffer() produce new ICE credentials even if
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// RTCOfferAnswerOptions::ice_restart is false.
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// https://w3c.github.io/webrtc-pc/#dfn-localufragstoreplace
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// TODO(hbos): When JsepTransportController/JsepTransport supports rollback,
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// move this type of logic to JsepTransportController/JsepTransport.
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class LocalIceCredentialsToReplace;
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struct RtpSenderInfo {
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RtpSenderInfo() : first_ssrc(0) {}
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RtpSenderInfo(const std::string& stream_id,
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const std::string sender_id,
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uint32_t ssrc)
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: stream_id(stream_id), sender_id(sender_id), first_ssrc(ssrc) {}
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bool operator==(const RtpSenderInfo& other) {
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return this->stream_id == other.stream_id &&
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this->sender_id == other.sender_id &&
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this->first_ssrc == other.first_ssrc;
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}
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std::string stream_id;
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std::string sender_id;
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// An RtpSender can have many SSRCs. The first one is used as a sort of ID
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// for communicating with the lower layers.
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uint32_t first_ssrc;
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};
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// Captures partial state to be used for rollback. Applicable only in
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// Unified Plan.
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class TransceiverStableState {
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public:
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TransceiverStableState() {}
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void set_newly_created();
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void SetMSectionIfUnset(absl::optional<std::string> mid,
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absl::optional<size_t> mline_index);
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void SetRemoteStreamIdsIfUnset(const std::vector<std::string>& ids);
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absl::optional<std::string> mid() const { return mid_; }
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absl::optional<size_t> mline_index() const { return mline_index_; }
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absl::optional<std::vector<std::string>> remote_stream_ids() const {
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return remote_stream_ids_;
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}
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bool has_m_section() const { return has_m_section_; }
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bool newly_created() const { return newly_created_; }
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private:
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absl::optional<std::string> mid_;
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absl::optional<size_t> mline_index_;
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absl::optional<std::vector<std::string>> remote_stream_ids_;
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// Indicates that mid value from stable state has been captured and
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// that rollback has to restore the transceiver. Also protects against
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// subsequent overwrites.
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bool has_m_section_ = false;
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// Indicates that the transceiver was created as part of applying a
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// description to track potential need for removing transceiver during
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// rollback.
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bool newly_created_ = false;
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};
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// Implements MessageHandler.
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void OnMessage(rtc::Message* msg) override;
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// Plan B helpers for getting the voice/video media channels for the single
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// audio/video transceiver, if it exists.
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cricket::VoiceMediaChannel* voice_media_channel() const
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RTC_RUN_ON(signaling_thread());
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cricket::VideoMediaChannel* video_media_channel() const
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RTC_RUN_ON(signaling_thread());
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std::vector<rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>>
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GetSendersInternal() const RTC_RUN_ON(signaling_thread());
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std::vector<
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rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>>
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GetReceiversInternal() const RTC_RUN_ON(signaling_thread());
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rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
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GetAudioTransceiver() const RTC_RUN_ON(signaling_thread());
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rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
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GetVideoTransceiver() const RTC_RUN_ON(signaling_thread());
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rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
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GetFirstAudioTransceiver() const RTC_RUN_ON(signaling_thread());
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// Implementation of the offer/answer exchange operations. These are chained
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// onto the |operations_chain_| when the public CreateOffer(), CreateAnswer(),
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// SetLocalDescription() and SetRemoteDescription() methods are invoked.
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void DoCreateOffer(
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const RTCOfferAnswerOptions& options,
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rtc::scoped_refptr<CreateSessionDescriptionObserver> observer);
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void DoCreateAnswer(
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const RTCOfferAnswerOptions& options,
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rtc::scoped_refptr<CreateSessionDescriptionObserver> observer);
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void DoSetLocalDescription(
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std::unique_ptr<SessionDescriptionInterface> desc,
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rtc::scoped_refptr<SetLocalDescriptionObserverInterface> observer);
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void DoSetRemoteDescription(
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std::unique_ptr<SessionDescriptionInterface> desc,
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rtc::scoped_refptr<SetRemoteDescriptionObserverInterface> observer);
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|
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void CreateAudioReceiver(MediaStreamInterface* stream,
|
|
const RtpSenderInfo& remote_sender_info)
|
|
RTC_RUN_ON(signaling_thread());
|
|
|
|
void CreateVideoReceiver(MediaStreamInterface* stream,
|
|
const RtpSenderInfo& remote_sender_info)
|
|
RTC_RUN_ON(signaling_thread());
|
|
rtc::scoped_refptr<RtpReceiverInterface> RemoveAndStopReceiver(
|
|
const RtpSenderInfo& remote_sender_info) RTC_RUN_ON(signaling_thread());
|
|
|
|
// May be called either by AddStream/RemoveStream, or when a track is
|
|
// added/removed from a stream previously added via AddStream.
|
|
void AddAudioTrack(AudioTrackInterface* track, MediaStreamInterface* stream)
|
|
RTC_RUN_ON(signaling_thread());
|
|
void RemoveAudioTrack(AudioTrackInterface* track,
|
|
MediaStreamInterface* stream)
|
|
RTC_RUN_ON(signaling_thread());
|
|
void AddVideoTrack(VideoTrackInterface* track, MediaStreamInterface* stream)
|
|
RTC_RUN_ON(signaling_thread());
|
|
void RemoveVideoTrack(VideoTrackInterface* track,
|
|
MediaStreamInterface* stream)
|
|
RTC_RUN_ON(signaling_thread());
|
|
|
|
// AddTrack implementation when Unified Plan is specified.
|
|
RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>> AddTrackUnifiedPlan(
|
|
rtc::scoped_refptr<MediaStreamTrackInterface> track,
|
|
const std::vector<std::string>& stream_ids)
|
|
RTC_RUN_ON(signaling_thread());
|
|
// AddTrack implementation when Plan B is specified.
|
|
RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>> AddTrackPlanB(
|
|
rtc::scoped_refptr<MediaStreamTrackInterface> track,
|
|
const std::vector<std::string>& stream_ids)
|
|
RTC_RUN_ON(signaling_thread());
|
|
|
|
// Returns the first RtpTransceiver suitable for a newly added track, if such
|
|
// transceiver is available.
|
|
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
|
FindFirstTransceiverForAddedTrack(
|
|
rtc::scoped_refptr<MediaStreamTrackInterface> track)
|
|
RTC_RUN_ON(signaling_thread());
|
|
|
|
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
|
FindTransceiverBySender(rtc::scoped_refptr<RtpSenderInterface> sender)
|
|
RTC_RUN_ON(signaling_thread());
|
|
|
|
// Internal implementation for AddTransceiver family of methods. If
|
|
// |fire_callback| is set, fires OnRenegotiationNeeded callback if successful.
|
|
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> AddTransceiver(
|
|
cricket::MediaType media_type,
|
|
rtc::scoped_refptr<MediaStreamTrackInterface> track,
|
|
const RtpTransceiverInit& init,
|
|
bool fire_callback = true) RTC_RUN_ON(signaling_thread());
|
|
|
|
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>
|
|
CreateSender(cricket::MediaType media_type,
|
|
const std::string& id,
|
|
rtc::scoped_refptr<MediaStreamTrackInterface> track,
|
|
const std::vector<std::string>& stream_ids,
|
|
const std::vector<RtpEncodingParameters>& send_encodings);
|
|
|
|
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
|
|
CreateReceiver(cricket::MediaType media_type, const std::string& receiver_id);
|
|
|
|
// Create a new RtpTransceiver of the given type and add it to the list of
|
|
// transceivers.
|
|
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
|
CreateAndAddTransceiver(
|
|
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> sender,
|
|
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
|
|
receiver) RTC_RUN_ON(signaling_thread());
|
|
|
|
void SetIceConnectionState(IceConnectionState new_state)
|
|
RTC_RUN_ON(signaling_thread());
|
|
void SetStandardizedIceConnectionState(
|
|
PeerConnectionInterface::IceConnectionState new_state)
|
|
RTC_RUN_ON(signaling_thread());
|
|
void SetConnectionState(
|
|
PeerConnectionInterface::PeerConnectionState new_state)
|
|
RTC_RUN_ON(signaling_thread());
|
|
|
|
// Called any time the IceGatheringState changes.
|
|
void OnIceGatheringChange(IceGatheringState new_state)
|
|
RTC_RUN_ON(signaling_thread());
|
|
// New ICE candidate has been gathered.
|
|
void OnIceCandidate(std::unique_ptr<IceCandidateInterface> candidate)
|
|
RTC_RUN_ON(signaling_thread());
|
|
// Gathering of an ICE candidate failed.
|
|
void OnIceCandidateError(const std::string& address,
|
|
int port,
|
|
const std::string& url,
|
|
int error_code,
|
|
const std::string& error_text)
|
|
RTC_RUN_ON(signaling_thread());
|
|
// Some local ICE candidates have been removed.
|
|
void OnIceCandidatesRemoved(const std::vector<cricket::Candidate>& candidates)
|
|
RTC_RUN_ON(signaling_thread());
|
|
|
|
void OnSelectedCandidatePairChanged(
|
|
const cricket::CandidatePairChangeEvent& event)
|
|
RTC_RUN_ON(signaling_thread());
|
|
|
|
// Update the state, signaling if necessary.
|
|
void ChangeSignalingState(SignalingState signaling_state)
|
|
RTC_RUN_ON(signaling_thread());
|
|
|
|
// Signals from MediaStreamObserver.
|
|
void OnAudioTrackAdded(AudioTrackInterface* track,
|
|
MediaStreamInterface* stream)
|
|
RTC_RUN_ON(signaling_thread());
|
|
void OnAudioTrackRemoved(AudioTrackInterface* track,
|
|
MediaStreamInterface* stream)
|
|
RTC_RUN_ON(signaling_thread());
|
|
void OnVideoTrackAdded(VideoTrackInterface* track,
|
|
MediaStreamInterface* stream)
|
|
RTC_RUN_ON(signaling_thread());
|
|
void OnVideoTrackRemoved(VideoTrackInterface* track,
|
|
MediaStreamInterface* stream)
|
|
RTC_RUN_ON(signaling_thread());
|
|
|
|
void PostSetSessionDescriptionSuccess(
|
|
SetSessionDescriptionObserver* observer);
|
|
void PostSetSessionDescriptionFailure(SetSessionDescriptionObserver* observer,
|
|
RTCError&& error);
|
|
void PostCreateSessionDescriptionFailure(
|
|
CreateSessionDescriptionObserver* observer,
|
|
RTCError error);
|
|
|
|
// Synchronous implementations of SetLocalDescription/SetRemoteDescription
|
|
// that return an RTCError instead of invoking a callback.
|
|
RTCError ApplyLocalDescription(
|
|
std::unique_ptr<SessionDescriptionInterface> desc);
|
|
RTCError ApplyRemoteDescription(
|
|
std::unique_ptr<SessionDescriptionInterface> desc);
|
|
|
|
// Updates the local RtpTransceivers according to the JSEP rules. Called as
|
|
// part of setting the local/remote description.
|
|
RTCError UpdateTransceiversAndDataChannels(
|
|
cricket::ContentSource source,
|
|
const SessionDescriptionInterface& new_session,
|
|
const SessionDescriptionInterface* old_local_description,
|
|
const SessionDescriptionInterface* old_remote_description)
|
|
RTC_RUN_ON(signaling_thread());
|
|
|
|
// Either creates or destroys the transceiver's BaseChannel according to the
|
|
// given media section.
|
|
RTCError UpdateTransceiverChannel(
|
|
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
|
transceiver,
|
|
const cricket::ContentInfo& content,
|
|
const cricket::ContentGroup* bundle_group) RTC_RUN_ON(signaling_thread());
|
|
|
|
// Either creates or destroys the local data channel according to the given
|
|
// media section.
|
|
RTCError UpdateDataChannel(cricket::ContentSource source,
|
|
const cricket::ContentInfo& content,
|
|
const cricket::ContentGroup* bundle_group)
|
|
RTC_RUN_ON(signaling_thread());
|
|
|
|
// Associate the given transceiver according to the JSEP rules.
|
|
RTCErrorOr<
|
|
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
|
|
AssociateTransceiver(cricket::ContentSource source,
|
|
SdpType type,
|
|
size_t mline_index,
|
|
const cricket::ContentInfo& content,
|
|
const cricket::ContentInfo* old_local_content,
|
|
const cricket::ContentInfo* old_remote_content)
|
|
RTC_RUN_ON(signaling_thread());
|
|
|
|
// Returns the RtpTransceiver, if found, that is associated to the given MID.
|
|
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
|
GetAssociatedTransceiver(const std::string& mid) const
|
|
RTC_RUN_ON(signaling_thread());
|
|
|
|
// Returns the RtpTransceiver, if found, that was assigned to the given mline
|
|
// index in CreateOffer.
|
|
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
|
GetTransceiverByMLineIndex(size_t mline_index) const
|
|
RTC_RUN_ON(signaling_thread());
|
|
|
|
// Returns an RtpTransciever, if available, that can be used to receive the
|
|
// given media type according to JSEP rules.
|
|
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
|
FindAvailableTransceiverToReceive(cricket::MediaType media_type) const
|
|
RTC_RUN_ON(signaling_thread());
|
|
|
|
// Returns the media section in the given session description that is
|
|
// associated with the RtpTransceiver. Returns null if none found or this
|
|
// RtpTransceiver is not associated. Logic varies depending on the
|
|
// SdpSemantics specified in the configuration.
|
|
const cricket::ContentInfo* FindMediaSectionForTransceiver(
|
|
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
|
transceiver,
|
|
const SessionDescriptionInterface* sdesc) const
|
|
RTC_RUN_ON(signaling_thread());
|
|
|
|
// Runs the algorithm **set the associated remote streams** specified in
|
|
// https://w3c.github.io/webrtc-pc/#set-associated-remote-streams.
|
|
void SetAssociatedRemoteStreams(
|
|
rtc::scoped_refptr<RtpReceiverInternal> receiver,
|
|
const std::vector<std::string>& stream_ids,
|
|
std::vector<rtc::scoped_refptr<MediaStreamInterface>>* added_streams,
|
|
std::vector<rtc::scoped_refptr<MediaStreamInterface>>* removed_streams)
|
|
RTC_RUN_ON(signaling_thread());
|
|
|
|
// Runs the algorithm **process the removal of a remote track** specified in
|
|
// the WebRTC specification.
|
|
// This method will update the following lists:
|
|
// |remove_list| is the list of transceivers for which the receiving track is
|
|
// being removed.
|
|
// |removed_streams| is the list of streams which no longer have a receiving
|
|
// track so should be removed.
|
|
// https://w3c.github.io/webrtc-pc/#process-remote-track-removal
|
|
void ProcessRemovalOfRemoteTrack(
|
|
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
|
transceiver,
|
|
std::vector<rtc::scoped_refptr<RtpTransceiverInterface>>* remove_list,
|
|
std::vector<rtc::scoped_refptr<MediaStreamInterface>>* removed_streams)
|
|
RTC_RUN_ON(signaling_thread());
|
|
|
|
void RemoveRemoteStreamsIfEmpty(
|
|
const std::vector<rtc::scoped_refptr<MediaStreamInterface>>&
|
|
remote_streams,
|
|
std::vector<rtc::scoped_refptr<MediaStreamInterface>>* removed_streams)
|
|
RTC_RUN_ON(signaling_thread());
|
|
|
|
void OnNegotiationNeeded();
|
|
|
|
// Returns a MediaSessionOptions struct with options decided by |options|,
|
|
// the local MediaStreams and DataChannels.
|
|
void GetOptionsForOffer(const PeerConnectionInterface::RTCOfferAnswerOptions&
|
|
offer_answer_options,
|
|
cricket::MediaSessionOptions* session_options)
|
|
RTC_RUN_ON(signaling_thread());
|
|
void GetOptionsForPlanBOffer(
|
|
const PeerConnectionInterface::RTCOfferAnswerOptions&
|
|
offer_answer_options,
|
|
cricket::MediaSessionOptions* session_options)
|
|
RTC_RUN_ON(signaling_thread());
|
|
void GetOptionsForUnifiedPlanOffer(
|
|
const PeerConnectionInterface::RTCOfferAnswerOptions&
|
|
offer_answer_options,
|
|
cricket::MediaSessionOptions* session_options)
|
|
RTC_RUN_ON(signaling_thread());
|
|
|
|
RTCError HandleLegacyOfferOptions(const RTCOfferAnswerOptions& options)
|
|
RTC_RUN_ON(signaling_thread());
|
|
void RemoveRecvDirectionFromReceivingTransceiversOfType(
|
|
cricket::MediaType media_type) RTC_RUN_ON(signaling_thread());
|
|
void AddUpToOneReceivingTransceiverOfType(cricket::MediaType media_type);
|
|
std::vector<
|
|
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
|
|
GetReceivingTransceiversOfType(cricket::MediaType media_type)
|
|
RTC_RUN_ON(signaling_thread());
|
|
|
|
// Returns a MediaSessionOptions struct with options decided by
|
|
// |constraints|, the local MediaStreams and DataChannels.
|
|
void GetOptionsForAnswer(const RTCOfferAnswerOptions& offer_answer_options,
|
|
cricket::MediaSessionOptions* session_options)
|
|
RTC_RUN_ON(signaling_thread());
|
|
void GetOptionsForPlanBAnswer(
|
|
const PeerConnectionInterface::RTCOfferAnswerOptions&
|
|
offer_answer_options,
|
|
cricket::MediaSessionOptions* session_options)
|
|
RTC_RUN_ON(signaling_thread());
|
|
void GetOptionsForUnifiedPlanAnswer(
|
|
const PeerConnectionInterface::RTCOfferAnswerOptions&
|
|
offer_answer_options,
|
|
cricket::MediaSessionOptions* session_options)
|
|
RTC_RUN_ON(signaling_thread());
|
|
|
|
// Generates MediaDescriptionOptions for the |session_opts| based on existing
|
|
// local description or remote description.
|
|
void GenerateMediaDescriptionOptions(
|
|
const SessionDescriptionInterface* session_desc,
|
|
RtpTransceiverDirection audio_direction,
|
|
RtpTransceiverDirection video_direction,
|
|
absl::optional<size_t>* audio_index,
|
|
absl::optional<size_t>* video_index,
|
|
absl::optional<size_t>* data_index,
|
|
cricket::MediaSessionOptions* session_options)
|
|
RTC_RUN_ON(signaling_thread());
|
|
|
|
// Generates the active MediaDescriptionOptions for the local data channel
|
|
// given the specified MID.
|
|
cricket::MediaDescriptionOptions GetMediaDescriptionOptionsForActiveData(
|
|
const std::string& mid) const RTC_RUN_ON(signaling_thread());
|
|
|
|
// Generates the rejected MediaDescriptionOptions for the local data channel
|
|
// given the specified MID.
|
|
cricket::MediaDescriptionOptions GetMediaDescriptionOptionsForRejectedData(
|
|
const std::string& mid) const RTC_RUN_ON(signaling_thread());
|
|
|
|
// Returns the MID for the data section associated with either the
|
|
// RtpDataChannel or SCTP data channel, if it has been set. If no data
|
|
// channels are configured this will return nullopt.
|
|
absl::optional<std::string> GetDataMid() const RTC_RUN_ON(signaling_thread());
|
|
|
|
// Remove all local and remote senders of type |media_type|.
|
|
// Called when a media type is rejected (m-line set to port 0).
|
|
void RemoveSenders(cricket::MediaType media_type)
|
|
RTC_RUN_ON(signaling_thread());
|
|
|
|
// Makes sure a MediaStreamTrack is created for each StreamParam in |streams|,
|
|
// and existing MediaStreamTracks are removed if there is no corresponding
|
|
// StreamParam. If |default_track_needed| is true, a default MediaStreamTrack
|
|
// is created if it doesn't exist; if false, it's removed if it exists.
|
|
// |media_type| is the type of the |streams| and can be either audio or video.
|
|
// If a new MediaStream is created it is added to |new_streams|.
|
|
void UpdateRemoteSendersList(
|
|
const std::vector<cricket::StreamParams>& streams,
|
|
bool default_track_needed,
|
|
cricket::MediaType media_type,
|
|
StreamCollection* new_streams) RTC_RUN_ON(signaling_thread());
|
|
|
|
// Triggered when a remote sender has been seen for the first time in a remote
|
|
// session description. It creates a remote MediaStreamTrackInterface
|
|
// implementation and triggers CreateAudioReceiver or CreateVideoReceiver.
|
|
void OnRemoteSenderAdded(const RtpSenderInfo& sender_info,
|
|
cricket::MediaType media_type)
|
|
RTC_RUN_ON(signaling_thread());
|
|
|
|
// Triggered when a remote sender has been removed from a remote session
|
|
// description. It removes the remote sender with id |sender_id| from a remote
|
|
// MediaStream and triggers DestroyAudioReceiver or DestroyVideoReceiver.
|
|
void OnRemoteSenderRemoved(const RtpSenderInfo& sender_info,
|
|
cricket::MediaType media_type)
|
|
RTC_RUN_ON(signaling_thread());
|
|
|
|
// Finds remote MediaStreams without any tracks and removes them from
|
|
// |remote_streams_| and notifies the observer that the MediaStreams no longer
|
|
// exist.
|
|
void UpdateEndedRemoteMediaStreams() RTC_RUN_ON(signaling_thread());
|
|
|
|
// Loops through the vector of |streams| and finds added and removed
|
|
// StreamParams since last time this method was called.
|
|
// For each new or removed StreamParam, OnLocalSenderSeen or
|
|
// OnLocalSenderRemoved is invoked.
|
|
void UpdateLocalSenders(const std::vector<cricket::StreamParams>& streams,
|
|
cricket::MediaType media_type)
|
|
RTC_RUN_ON(signaling_thread());
|
|
|
|
// Triggered when a local sender has been seen for the first time in a local
|
|
// session description.
|
|
// This method triggers CreateAudioSender or CreateVideoSender if the rtp
|
|
// streams in the local SessionDescription can be mapped to a MediaStreamTrack
|
|
// in a MediaStream in |local_streams_|
|
|
void OnLocalSenderAdded(const RtpSenderInfo& sender_info,
|
|
cricket::MediaType media_type)
|
|
RTC_RUN_ON(signaling_thread());
|
|
|
|
// Triggered when a local sender has been removed from a local session
|
|
// description.
|
|
// This method triggers DestroyAudioSender or DestroyVideoSender if a stream
|
|
// has been removed from the local SessionDescription and the stream can be
|
|
// mapped to a MediaStreamTrack in a MediaStream in |local_streams_|.
|
|
void OnLocalSenderRemoved(const RtpSenderInfo& sender_info,
|
|
cricket::MediaType media_type)
|
|
RTC_RUN_ON(signaling_thread());
|
|
|
|
// Returns true if the PeerConnection is configured to use Unified Plan
|
|
// semantics for creating offers/answers and setting local/remote
|
|
// descriptions. If this is true the RtpTransceiver API will also be available
|
|
// to the user. If this is false, Plan B semantics are assumed.
|
|
// TODO(bugs.webrtc.org/8530): Flip the default to be Unified Plan once
|
|
// sufficient time has passed.
|
|
bool IsUnifiedPlan() const RTC_RUN_ON(signaling_thread()) {
|
|
return configuration_.sdp_semantics == SdpSemantics::kUnifiedPlan;
|
|
}
|
|
|
|
// The offer/answer machinery assumes the media section MID is present and
|
|
// unique. To support legacy end points that do not supply a=mid lines, this
|
|
// method will modify the session description to add MIDs generated according
|
|
// to the SDP semantics.
|
|
void FillInMissingRemoteMids(cricket::SessionDescription* remote_description)
|
|
RTC_RUN_ON(signaling_thread());
|
|
|
|
// Return the RtpSender with the given track attached.
|
|
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>
|
|
FindSenderForTrack(MediaStreamTrackInterface* track) const
|
|
RTC_RUN_ON(signaling_thread());
|
|
|
|
// Return the RtpSender with the given id, or null if none exists.
|
|
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>
|
|
FindSenderById(const std::string& sender_id) const
|
|
RTC_RUN_ON(signaling_thread());
|
|
|
|
// Return the RtpReceiver with the given id, or null if none exists.
|
|
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
|
|
FindReceiverById(const std::string& receiver_id) const
|
|
RTC_RUN_ON(signaling_thread());
|
|
|
|
std::vector<RtpSenderInfo>* GetRemoteSenderInfos(
|
|
cricket::MediaType media_type);
|
|
std::vector<RtpSenderInfo>* GetLocalSenderInfos(
|
|
cricket::MediaType media_type);
|
|
const RtpSenderInfo* FindSenderInfo(const std::vector<RtpSenderInfo>& infos,
|
|
const std::string& stream_id,
|
|
const std::string sender_id) const;
|
|
|
|
// Returns the specified SCTP DataChannel in sctp_data_channels_,
|
|
// or nullptr if not found.
|
|
SctpDataChannel* FindDataChannelBySid(int sid) const
|
|
RTC_RUN_ON(signaling_thread());
|
|
|
|
// Called when first configuring the port allocator.
|
|
struct InitializePortAllocatorResult {
|
|
bool enable_ipv6;
|
|
};
|
|
InitializePortAllocatorResult InitializePortAllocator_n(
|
|
const cricket::ServerAddresses& stun_servers,
|
|
const std::vector<cricket::RelayServerConfig>& turn_servers,
|
|
const RTCConfiguration& configuration);
|
|
// Called when SetConfiguration is called to apply the supported subset
|
|
// of the configuration on the network thread.
|
|
bool ReconfigurePortAllocator_n(
|
|
const cricket::ServerAddresses& stun_servers,
|
|
const std::vector<cricket::RelayServerConfig>& turn_servers,
|
|
IceTransportsType type,
|
|
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.
|
|
// This function should only be called from the worker thread.
|
|
bool StartRtcEventLog_w(std::unique_ptr<RtcEventLogOutput> output,
|
|
int64_t output_period_ms);
|
|
|
|
// Stops recording an RTC event log.
|
|
// This function should only be called from the worker thread.
|
|
void StopRtcEventLog_w();
|
|
|
|
// Ensures the configuration doesn't have any parameters with invalid values,
|
|
// or values that conflict with other parameters.
|
|
//
|
|
// Returns RTCError::OK() if there are no issues.
|
|
RTCError ValidateConfiguration(const RTCConfiguration& config) const;
|
|
|
|
cricket::ChannelManager* channel_manager() const;
|
|
|
|
enum class SessionError {
|
|
kNone, // No error.
|
|
kContent, // Error in BaseChannel SetLocalContent/SetRemoteContent.
|
|
kTransport, // Error from the underlying transport.
|
|
};
|
|
|
|
// Returns the last error in the session. See the enum above for details.
|
|
SessionError session_error() const RTC_RUN_ON(signaling_thread()) {
|
|
return session_error_;
|
|
}
|
|
const std::string& session_error_desc() const { return session_error_desc_; }
|
|
|
|
cricket::ChannelInterface* GetChannel(const std::string& content_name);
|
|
|
|
cricket::IceConfig ParseIceConfig(
|
|
const PeerConnectionInterface::RTCConfiguration& config) const;
|
|
|
|
cricket::DataChannelType data_channel_type() const;
|
|
|
|
// Called when an RTCCertificate is generated or retrieved by
|
|
// WebRTCSessionDescriptionFactory. Should happen before setLocalDescription.
|
|
void OnCertificateReady(
|
|
const rtc::scoped_refptr<rtc::RTCCertificate>& certificate);
|
|
void OnDtlsSrtpSetupFailure(cricket::BaseChannel*, bool rtcp);
|
|
|
|
// Non-const versions of local_description()/remote_description(), for use
|
|
// internally.
|
|
SessionDescriptionInterface* mutable_local_description()
|
|
RTC_RUN_ON(signaling_thread()) {
|
|
return pending_local_description_ ? pending_local_description_.get()
|
|
: current_local_description_.get();
|
|
}
|
|
SessionDescriptionInterface* mutable_remote_description()
|
|
RTC_RUN_ON(signaling_thread()) {
|
|
return pending_remote_description_ ? pending_remote_description_.get()
|
|
: current_remote_description_.get();
|
|
}
|
|
|
|
// Updates the error state, signaling if necessary.
|
|
void SetSessionError(SessionError error, const std::string& error_desc);
|
|
|
|
RTCError UpdateSessionState(SdpType type,
|
|
cricket::ContentSource source,
|
|
const cricket::SessionDescription* description);
|
|
// Push the media parts of the local or remote session description
|
|
// down to all of the channels.
|
|
RTCError PushdownMediaDescription(SdpType type, cricket::ContentSource source)
|
|
RTC_RUN_ON(signaling_thread());
|
|
|
|
RTCError PushdownTransportDescription(cricket::ContentSource source,
|
|
SdpType type);
|
|
|
|
// Returns true and the TransportInfo of the given |content_name|
|
|
// from |description|. Returns false if it's not available.
|
|
static bool GetTransportDescription(
|
|
const cricket::SessionDescription* description,
|
|
const std::string& content_name,
|
|
cricket::TransportDescription* info);
|
|
|
|
// Enables media channels to allow sending of media.
|
|
// This enables media to flow on all configured audio/video channels and the
|
|
// RtpDataChannel.
|
|
void EnableSending() RTC_RUN_ON(signaling_thread());
|
|
|
|
// Destroys all BaseChannels and destroys the SCTP data channel, if present.
|
|
void DestroyAllChannels() RTC_RUN_ON(signaling_thread());
|
|
|
|
// 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());
|
|
// Uses all remote candidates in |remote_desc| in this session.
|
|
bool UseCandidatesInSessionDescription(
|
|
const SessionDescriptionInterface* remote_desc)
|
|
RTC_RUN_ON(signaling_thread());
|
|
// Uses |candidate| in this session.
|
|
bool UseCandidate(const IceCandidateInterface* candidate)
|
|
RTC_RUN_ON(signaling_thread());
|
|
RTCErrorOr<const cricket::ContentInfo*> FindContentInfo(
|
|
const SessionDescriptionInterface* description,
|
|
const IceCandidateInterface* candidate) RTC_RUN_ON(signaling_thread());
|
|
// Deletes the corresponding channel of contents that don't exist in |desc|.
|
|
// |desc| can be null. This means that all channels are deleted.
|
|
void RemoveUnusedChannels(const cricket::SessionDescription* desc)
|
|
RTC_RUN_ON(signaling_thread());
|
|
|
|
// Allocates media channels based on the |desc|. If |desc| doesn't have
|
|
// the BUNDLE option, this method will disable BUNDLE in PortAllocator.
|
|
// This method will also delete any existing media channels before creating.
|
|
RTCError CreateChannels(const cricket::SessionDescription& desc)
|
|
RTC_RUN_ON(signaling_thread());
|
|
|
|
// If the BUNDLE policy is max-bundle, then we know for sure that all
|
|
// transports will be bundled from the start. This method returns the BUNDLE
|
|
// group if that's the case, or null if BUNDLE will be negotiated later. An
|
|
// error is returned if max-bundle is specified but the session description
|
|
// does not have a BUNDLE group.
|
|
RTCErrorOr<const cricket::ContentGroup*> GetEarlyBundleGroup(
|
|
const cricket::SessionDescription& desc) const
|
|
RTC_RUN_ON(signaling_thread());
|
|
|
|
// Helper methods to create media channels.
|
|
cricket::VoiceChannel* CreateVoiceChannel(const std::string& mid)
|
|
RTC_RUN_ON(signaling_thread());
|
|
cricket::VideoChannel* CreateVideoChannel(const std::string& mid)
|
|
RTC_RUN_ON(signaling_thread());
|
|
bool CreateDataChannel(const std::string& mid) RTC_RUN_ON(signaling_thread());
|
|
|
|
bool SetupDataChannelTransport_n(const std::string& mid)
|
|
RTC_RUN_ON(network_thread());
|
|
void TeardownDataChannelTransport_n() RTC_RUN_ON(network_thread());
|
|
|
|
bool ValidateBundleSettings(const cricket::SessionDescription* desc);
|
|
bool HasRtcpMuxEnabled(const cricket::ContentInfo* content);
|
|
// Below methods are helper methods which verifies SDP.
|
|
RTCError ValidateSessionDescription(const SessionDescriptionInterface* sdesc,
|
|
cricket::ContentSource source)
|
|
RTC_RUN_ON(signaling_thread());
|
|
|
|
// Check if a call to SetLocalDescription is acceptable with a session
|
|
// description of the given type.
|
|
bool ExpectSetLocalDescription(SdpType type);
|
|
// Check if a call to SetRemoteDescription is acceptable with a session
|
|
// description of the given type.
|
|
bool ExpectSetRemoteDescription(SdpType type);
|
|
// Verifies a=setup attribute as per RFC 5763.
|
|
bool ValidateDtlsSetupAttribute(const cricket::SessionDescription* desc,
|
|
SdpType type);
|
|
|
|
// Returns true if we are ready to push down the remote candidate.
|
|
// |remote_desc| is the new remote description, or NULL if the current remote
|
|
// description should be used. Output |valid| is true if the candidate media
|
|
// index is valid.
|
|
bool ReadyToUseRemoteCandidate(const IceCandidateInterface* candidate,
|
|
const SessionDescriptionInterface* remote_desc,
|
|
bool* valid) RTC_RUN_ON(signaling_thread());
|
|
|
|
// Returns true if SRTP (either using DTLS-SRTP or SDES) is required by
|
|
// this session.
|
|
bool SrtpRequired() const RTC_RUN_ON(signaling_thread());
|
|
|
|
// 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);
|
|
|
|
const char* SessionErrorToString(SessionError error) const;
|
|
std::string GetSessionErrorMsg() RTC_RUN_ON(signaling_thread());
|
|
|
|
// Report the UMA metric SdpFormatReceived for the given remote offer.
|
|
void ReportSdpFormatReceived(const SessionDescriptionInterface& remote_offer);
|
|
|
|
// Report inferred negotiated SDP semantics from a local/remote answer to the
|
|
// UMA observer.
|
|
void ReportNegotiatedSdpSemantics(const SessionDescriptionInterface& answer);
|
|
|
|
// Invoked when TransportController connection completion is signaled.
|
|
// Reports stats for all transports in use.
|
|
void ReportTransportStats() RTC_RUN_ON(signaling_thread());
|
|
|
|
// Gather the usage of IPv4/IPv6 as best connection.
|
|
void ReportBestConnectionState(const cricket::TransportStats& stats);
|
|
|
|
void ReportNegotiatedCiphers(const cricket::TransportStats& stats,
|
|
const std::set<cricket::MediaType>& media_types)
|
|
RTC_RUN_ON(signaling_thread());
|
|
void ReportIceCandidateCollected(const cricket::Candidate& candidate)
|
|
RTC_RUN_ON(signaling_thread());
|
|
void ReportRemoteIceCandidateAdded(const cricket::Candidate& candidate)
|
|
RTC_RUN_ON(signaling_thread());
|
|
|
|
void NoteUsageEvent(UsageEvent event);
|
|
void ReportUsagePattern() const RTC_RUN_ON(signaling_thread());
|
|
|
|
void OnSentPacket_w(const rtc::SentPacket& sent_packet);
|
|
|
|
const std::string GetTransportName(const std::string& content_name)
|
|
RTC_RUN_ON(signaling_thread());
|
|
|
|
// Functions for dealing with transports.
|
|
// Note that cricket code uses the term "channel" for what other code
|
|
// refers to as "transport".
|
|
|
|
// Destroys and clears the BaseChannel associated with the given transceiver,
|
|
// if such channel is set.
|
|
void DestroyTransceiverChannel(
|
|
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
|
transceiver);
|
|
|
|
// Destroys the RTP data channel transport and/or the SCTP data channel
|
|
// transport and clears it.
|
|
void DestroyDataChannelTransport() RTC_RUN_ON(signaling_thread());
|
|
|
|
// Destroys the given ChannelInterface.
|
|
// The channel cannot be accessed after this method is called.
|
|
void DestroyChannelInterface(cricket::ChannelInterface* channel);
|
|
|
|
// 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;
|
|
|
|
// RtpSenderBase::SetStreamsObserver override.
|
|
void OnSetStreams() override;
|
|
|
|
// Returns the CryptoOptions for this PeerConnection. This will always
|
|
// return the RTCConfiguration.crypto_options if set and will only default
|
|
// back to the PeerConnectionFactory settings if nothing was set.
|
|
CryptoOptions GetCryptoOptions() RTC_RUN_ON(signaling_thread());
|
|
|
|
// Returns rtp transport, result can not be nullptr.
|
|
RtpTransportInternal* GetRtpTransport(const std::string& mid)
|
|
RTC_RUN_ON(signaling_thread()) {
|
|
auto rtp_transport = transport_controller_->GetRtpTransport(mid);
|
|
RTC_DCHECK(rtp_transport);
|
|
return rtp_transport;
|
|
}
|
|
|
|
void UpdateNegotiationNeeded();
|
|
bool CheckIfNegotiationIsNeeded();
|
|
|
|
// | sdp_type | is the type of the SDP that caused the rollback.
|
|
RTCError Rollback(SdpType sdp_type);
|
|
|
|
// Storing the factory as a scoped reference pointer ensures that the memory
|
|
// in the PeerConnectionFactoryImpl remains available as long as the
|
|
// PeerConnection is running. It is passed to PeerConnection as a raw pointer.
|
|
// However, since the reference counting is done in the
|
|
// PeerConnectionFactoryInterface all instances created using the raw pointer
|
|
// will refer to the same reference count.
|
|
const rtc::scoped_refptr<PeerConnectionFactory> factory_;
|
|
PeerConnectionObserver* observer_ RTC_GUARDED_BY(signaling_thread()) =
|
|
nullptr;
|
|
|
|
// 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());
|
|
|
|
// The operations chain is used by the offer/answer exchange methods to ensure
|
|
// they are executed in the right order. For example, if
|
|
// SetRemoteDescription() is invoked while CreateOffer() is still pending, the
|
|
// SRD operation will not start until CreateOffer() has completed. See
|
|
// https://w3c.github.io/webrtc-pc/#dfn-operations-chain.
|
|
rtc::scoped_refptr<rtc::OperationsChain> operations_chain_
|
|
RTC_GUARDED_BY(signaling_thread());
|
|
|
|
SignalingState signaling_state_ RTC_GUARDED_BY(signaling_thread()) = kStable;
|
|
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());
|
|
|
|
// TODO(zstein): |async_resolver_factory_| can currently be nullptr if it
|
|
// is not injected. It should be required once chromium supplies it.
|
|
std::unique_ptr<AsyncResolverFactory> async_resolver_factory_
|
|
RTC_GUARDED_BY(signaling_thread());
|
|
std::unique_ptr<cricket::PortAllocator>
|
|
port_allocator_; // TODO(bugs.webrtc.org/9987): Accessed on both
|
|
// signaling and network thread.
|
|
std::unique_ptr<rtc::PacketSocketFactory> packet_socket_factory_;
|
|
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.
|
|
std::unique_ptr<rtc::SSLCertificateVerifier>
|
|
tls_cert_verifier_; // TODO(bugs.webrtc.org/9987): Accessed on both
|
|
// signaling and network thread.
|
|
|
|
// One PeerConnection has only one RTCP CNAME.
|
|
// https://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-26#section-4.9
|
|
const std::string rtcp_cname_;
|
|
|
|
// Streams added via AddStream.
|
|
const rtc::scoped_refptr<StreamCollection> local_streams_
|
|
RTC_GUARDED_BY(signaling_thread());
|
|
// Streams created as a result of SetRemoteDescription.
|
|
const rtc::scoped_refptr<StreamCollection> remote_streams_
|
|
RTC_GUARDED_BY(signaling_thread());
|
|
|
|
std::vector<std::unique_ptr<MediaStreamObserver>> stream_observers_
|
|
RTC_GUARDED_BY(signaling_thread());
|
|
|
|
// These lists store sender info seen in local/remote descriptions.
|
|
std::vector<RtpSenderInfo> remote_audio_sender_infos_
|
|
RTC_GUARDED_BY(signaling_thread());
|
|
std::vector<RtpSenderInfo> remote_video_sender_infos_
|
|
RTC_GUARDED_BY(signaling_thread());
|
|
std::vector<RtpSenderInfo> local_audio_sender_infos_
|
|
RTC_GUARDED_BY(signaling_thread());
|
|
std::vector<RtpSenderInfo> local_video_sender_infos_
|
|
RTC_GUARDED_BY(signaling_thread());
|
|
|
|
bool remote_peer_supports_msid_ RTC_GUARDED_BY(signaling_thread()) = false;
|
|
|
|
// 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());
|
|
|
|
rtc::AsyncInvoker rtcp_invoker_ RTC_GUARDED_BY(network_thread());
|
|
|
|
// Points to the same thing as `call_`. Since it's const, we may read the
|
|
// pointer from any thread.
|
|
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());
|
|
// Holds changes made to transceivers during applying descriptors for
|
|
// potential rollback. Gets cleared once signaling state goes to stable.
|
|
std::map<rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>,
|
|
TransceiverStableState>
|
|
transceiver_stable_states_by_transceivers_;
|
|
// Used when rolling back RTP data channels.
|
|
bool have_pending_rtp_data_channel_ RTC_GUARDED_BY(signaling_thread()) =
|
|
false;
|
|
// Holds remote stream ids for transceivers from stable state.
|
|
std::map<rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>,
|
|
std::vector<std::string>>
|
|
remote_stream_ids_by_transceivers_;
|
|
std::vector<
|
|
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
|
|
transceivers_; // TODO(bugs.webrtc.org/9987): Accessed on both signaling
|
|
// and network thread.
|
|
|
|
// In Unified Plan, if we encounter remote SDP that does not contain an a=msid
|
|
// line we create and use a stream with a random ID for our receivers. This is
|
|
// to support legacy endpoints that do not support the a=msid attribute (as
|
|
// opposed to streamless tracks with "a=msid:-").
|
|
rtc::scoped_refptr<MediaStreamInterface> missing_msid_default_stream_
|
|
RTC_GUARDED_BY(signaling_thread());
|
|
// MIDs will be generated using this generator which will keep track of
|
|
// all the MIDs that have been seen over the life of the PeerConnection.
|
|
rtc::UniqueStringGenerator mid_generator_ RTC_GUARDED_BY(signaling_thread());
|
|
|
|
SessionError session_error_ RTC_GUARDED_BY(signaling_thread()) =
|
|
SessionError::kNone;
|
|
std::string session_error_desc_ RTC_GUARDED_BY(signaling_thread());
|
|
|
|
std::string session_id_ RTC_GUARDED_BY(signaling_thread());
|
|
|
|
std::unique_ptr<JsepTransportController>
|
|
transport_controller_; // TODO(bugs.webrtc.org/9987): Accessed on both
|
|
// signaling and network thread.
|
|
std::unique_ptr<cricket::SctpTransportInternalFactory>
|
|
sctp_factory_; // 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());
|
|
|
|
// Whether this peer is the caller. Set when the local description is applied.
|
|
absl::optional<bool> is_caller_ RTC_GUARDED_BY(signaling_thread());
|
|
|
|
|
|
|
|
std::unique_ptr<SessionDescriptionInterface> current_local_description_
|
|
RTC_GUARDED_BY(signaling_thread());
|
|
std::unique_ptr<SessionDescriptionInterface> pending_local_description_
|
|
RTC_GUARDED_BY(signaling_thread());
|
|
std::unique_ptr<SessionDescriptionInterface> current_remote_description_
|
|
RTC_GUARDED_BY(signaling_thread());
|
|
std::unique_ptr<SessionDescriptionInterface> pending_remote_description_
|
|
RTC_GUARDED_BY(signaling_thread());
|
|
bool dtls_enabled_ RTC_GUARDED_BY(signaling_thread()) = false;
|
|
|
|
// List of content names for which the remote side triggered an ICE restart.
|
|
std::set<std::string> pending_ice_restarts_
|
|
RTC_GUARDED_BY(signaling_thread());
|
|
|
|
std::unique_ptr<WebRtcSessionDescriptionFactory> webrtc_session_desc_factory_
|
|
RTC_GUARDED_BY(signaling_thread());
|
|
|
|
// Member variables for caching global options.
|
|
cricket::AudioOptions audio_options_ RTC_GUARDED_BY(signaling_thread());
|
|
cricket::VideoOptions video_options_ RTC_GUARDED_BY(signaling_thread());
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int usage_event_accumulator_ RTC_GUARDED_BY(signaling_thread()) = 0;
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bool return_histogram_very_quickly_ RTC_GUARDED_BY(signaling_thread()) =
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false;
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// This object should be used to generate any SSRC that is not explicitly
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// specified by the user (or by the remote party).
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// The generator is not used directly, instead it is passed on to the
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// channel manager and the session description factory.
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rtc::UniqueRandomIdGenerator ssrc_generator_
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RTC_GUARDED_BY(signaling_thread());
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// A video bitrate allocator factory.
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// This can injected using the PeerConnectionDependencies,
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// or else the CreateBuiltinVideoBitrateAllocatorFactory() will be called.
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// Note that one can still choose to override this in a MediaEngine
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// if one wants too.
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std::unique_ptr<webrtc::VideoBitrateAllocatorFactory>
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video_bitrate_allocator_factory_;
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std::unique_ptr<LocalIceCredentialsToReplace>
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local_ice_credentials_to_replace_ RTC_GUARDED_BY(signaling_thread());
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bool is_negotiation_needed_ RTC_GUARDED_BY(signaling_thread()) = false;
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DataChannelController data_channel_controller_;
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rtc::WeakPtrFactory<PeerConnection> weak_ptr_factory_
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RTC_GUARDED_BY(signaling_thread());
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};
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} // namespace webrtc
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#endif // PC_PEER_CONNECTION_H_
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