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Nagram/TMessagesProj/jni/webrtc/pc/peer_connection.h

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Update to 7.0.0 (2060)
2020-08-14 16:58:22 +00:00
/*
* Copyright 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef PC_PEER_CONNECTION_H_
#define PC_PEER_CONNECTION_H_
#include <map>
#include <memory>
#include <set>
#include <string>
#include <utility>
#include <vector>
#include "api/peer_connection_interface.h"
#include "api/transport/data_channel_transport_interface.h"
#include "api/turn_customizer.h"
#include "pc/data_channel_controller.h"
#include "pc/ice_server_parsing.h"
#include "pc/jsep_transport_controller.h"
#include "pc/peer_connection_factory.h"
#include "pc/peer_connection_internal.h"
#include "pc/rtc_stats_collector.h"
#include "pc/rtp_sender.h"
#include "pc/rtp_transceiver.h"
#include "pc/sctp_transport.h"
#include "pc/stats_collector.h"
#include "pc/stream_collection.h"
#include "pc/webrtc_session_description_factory.h"
#include "rtc_base/experiments/field_trial_parser.h"
#include "rtc_base/operations_chain.h"
#include "rtc_base/race_checker.h"
#include "rtc_base/unique_id_generator.h"
#include "rtc_base/weak_ptr.h"
namespace webrtc {
class MediaStreamObserver;
class VideoRtpReceiver;
class RtcEventLog;
// PeerConnection is the implementation of the PeerConnection object as defined
// by the PeerConnectionInterface API surface.
// The class currently is solely responsible for the following:
// - Managing the session state machine (signaling state).
// - Creating and initializing lower-level objects, like PortAllocator and
// BaseChannels.
// - Owning and managing the life cycle of the RtpSender/RtpReceiver and track
// objects.
// - Tracking the current and pending local/remote session descriptions.
// The class currently is jointly responsible for the following:
// - Parsing and interpreting SDP.
// - Generating offers and answers based on the current state.
// - The ICE state machine.
// - Generating stats.
class PeerConnection : public PeerConnectionInternal,
public JsepTransportController::Observer,
public RtpSenderBase::SetStreamsObserver,
public rtc::MessageHandler,
public sigslot::has_slots<> {
public:
// A bit in the usage pattern is registered when its defining event occurs at
// least once.
enum class UsageEvent : int {
TURN_SERVER_ADDED = 0x01,
STUN_SERVER_ADDED = 0x02,
DATA_ADDED = 0x04,
AUDIO_ADDED = 0x08,
VIDEO_ADDED = 0x10,
// |SetLocalDescription| returns successfully.
SET_LOCAL_DESCRIPTION_SUCCEEDED = 0x20,
// |SetRemoteDescription| returns successfully.
SET_REMOTE_DESCRIPTION_SUCCEEDED = 0x40,
// A local candidate (with type host, server-reflexive, or relay) is
// collected.
CANDIDATE_COLLECTED = 0x80,
// A remote candidate is successfully added via |AddIceCandidate|.
ADD_ICE_CANDIDATE_SUCCEEDED = 0x100,
ICE_STATE_CONNECTED = 0x200,
CLOSE_CALLED = 0x400,
// A local candidate with private IP is collected.
PRIVATE_CANDIDATE_COLLECTED = 0x800,
// A remote candidate with private IP is added, either via AddiceCandidate
// or from the remote description.
REMOTE_PRIVATE_CANDIDATE_ADDED = 0x1000,
// A local mDNS candidate is collected.
MDNS_CANDIDATE_COLLECTED = 0x2000,
// A remote mDNS candidate is added, either via AddIceCandidate or from the
// remote description.
REMOTE_MDNS_CANDIDATE_ADDED = 0x4000,
// A local candidate with IPv6 address is collected.
IPV6_CANDIDATE_COLLECTED = 0x8000,
// A remote candidate with IPv6 address is added, either via AddIceCandidate
// or from the remote description.
REMOTE_IPV6_CANDIDATE_ADDED = 0x10000,
// A remote candidate (with type host, server-reflexive, or relay) is
// successfully added, either via AddIceCandidate or from the remote
// description.
REMOTE_CANDIDATE_ADDED = 0x20000,
// An explicit host-host candidate pair is selected, i.e. both the local and
// the remote candidates have the host type. This does not include candidate
// pairs formed with equivalent prflx remote candidates, e.g. a host-prflx
// pair where the prflx candidate has the same base as a host candidate of
// the remote peer.
DIRECT_CONNECTION_SELECTED = 0x40000,
MAX_VALUE = 0x80000,
};
explicit PeerConnection(PeerConnectionFactory* factory,
std::unique_ptr<RtcEventLog> event_log,
std::unique_ptr<Call> call);
bool Initialize(
const PeerConnectionInterface::RTCConfiguration& configuration,
PeerConnectionDependencies dependencies);
rtc::scoped_refptr<StreamCollectionInterface> local_streams() override;
rtc::scoped_refptr<StreamCollectionInterface> remote_streams() override;
bool AddStream(MediaStreamInterface* local_stream) override;
void RemoveStream(MediaStreamInterface* local_stream) override;
RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>> AddTrack(
rtc::scoped_refptr<MediaStreamTrackInterface> track,
const std::vector<std::string>& stream_ids) override;
bool RemoveTrack(RtpSenderInterface* sender) override;
RTCError RemoveTrackNew(
rtc::scoped_refptr<RtpSenderInterface> sender) override;
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> AddTransceiver(
rtc::scoped_refptr<MediaStreamTrackInterface> track) override;
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> AddTransceiver(
rtc::scoped_refptr<MediaStreamTrackInterface> track,
const RtpTransceiverInit& init) override;
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> AddTransceiver(
cricket::MediaType media_type) override;
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> AddTransceiver(
cricket::MediaType media_type,
const RtpTransceiverInit& init) override;
// Gets the DTLS SSL certificate associated with the audio transport on the
// remote side. This will become populated once the DTLS connection with the
// peer has been completed, as indicated by the ICE connection state
// transitioning to kIceConnectionCompleted.
// Note that this will be removed once we implement RTCDtlsTransport which
// has standardized method for getting this information.
// See https://www.w3.org/TR/webrtc/#rtcdtlstransport-interface
std::unique_ptr<rtc::SSLCertificate> GetRemoteAudioSSLCertificate();
// Version of the above method that returns the full certificate chain.
std::unique_ptr<rtc::SSLCertChain> GetRemoteAudioSSLCertChain();
rtc::scoped_refptr<RtpSenderInterface> CreateSender(
const std::string& kind,
const std::string& stream_id) override;
std::vector<rtc::scoped_refptr<RtpSenderInterface>> GetSenders()
const override;
std::vector<rtc::scoped_refptr<RtpReceiverInterface>> GetReceivers()
const override;
std::vector<rtc::scoped_refptr<RtpTransceiverInterface>> GetTransceivers()
const override;
rtc::scoped_refptr<DataChannelInterface> CreateDataChannel(
const std::string& label,
const DataChannelInit* config) override;
// WARNING: LEGACY. See peerconnectioninterface.h
bool GetStats(StatsObserver* observer,
webrtc::MediaStreamTrackInterface* track,
StatsOutputLevel level) override;
// Spec-complaint GetStats(). See peerconnectioninterface.h
void GetStats(RTCStatsCollectorCallback* callback) override;
void GetStats(
rtc::scoped_refptr<RtpSenderInterface> selector,
rtc::scoped_refptr<RTCStatsCollectorCallback> callback) override;
void GetStats(
rtc::scoped_refptr<RtpReceiverInterface> selector,
rtc::scoped_refptr<RTCStatsCollectorCallback> callback) override;
void ClearStatsCache() override;
SignalingState signaling_state() override;
IceConnectionState ice_connection_state() override;
IceConnectionState standardized_ice_connection_state() override;
PeerConnectionState peer_connection_state() override;
IceGatheringState ice_gathering_state() override;
absl::optional<bool> can_trickle_ice_candidates() override;
const SessionDescriptionInterface* local_description() const override;
const SessionDescriptionInterface* remote_description() const override;
const SessionDescriptionInterface* current_local_description() const override;
const SessionDescriptionInterface* current_remote_description()
const override;
const SessionDescriptionInterface* pending_local_description() const override;
const SessionDescriptionInterface* pending_remote_description()
const override;
void RestartIce() override;
// JSEP01
void CreateOffer(CreateSessionDescriptionObserver* observer,
const RTCOfferAnswerOptions& options) override;
void CreateAnswer(CreateSessionDescriptionObserver* observer,
const RTCOfferAnswerOptions& options) override;
void SetLocalDescription(
std::unique_ptr<SessionDescriptionInterface> desc,
rtc::scoped_refptr<SetLocalDescriptionObserverInterface> observer)
override;
void SetLocalDescription(
rtc::scoped_refptr<SetLocalDescriptionObserverInterface> observer)
override;
// TODO(https://crbug.com/webrtc/11798): Delete these methods in favor of the
// ones taking SetLocalDescriptionObserverInterface as argument.
void SetLocalDescription(SetSessionDescriptionObserver* observer,
SessionDescriptionInterface* desc) override;
void SetLocalDescription(SetSessionDescriptionObserver* observer) override;
void SetRemoteDescription(
std::unique_ptr<SessionDescriptionInterface> desc,
rtc::scoped_refptr<SetRemoteDescriptionObserverInterface> observer)
override;
// TODO(https://crbug.com/webrtc/11798): Delete this methods in favor of the
// ones taking SetRemoteDescriptionObserverInterface as argument.
void SetRemoteDescription(SetSessionDescriptionObserver* observer,
SessionDescriptionInterface* desc) override;
PeerConnectionInterface::RTCConfiguration GetConfiguration() override;
RTCError SetConfiguration(
const PeerConnectionInterface::RTCConfiguration& configuration) override;
bool AddIceCandidate(const IceCandidateInterface* candidate) override;
void AddIceCandidate(std::unique_ptr<IceCandidateInterface> candidate,
std::function<void(RTCError)> callback) override;
bool RemoveIceCandidates(
const std::vector<cricket::Candidate>& candidates) override;
RTCError SetBitrate(const BitrateSettings& bitrate) override;
void SetAudioPlayout(bool playout) override;
void SetAudioRecording(bool recording) override;
rtc::scoped_refptr<DtlsTransportInterface> LookupDtlsTransportByMid(
const std::string& mid) override;
rtc::scoped_refptr<DtlsTransport> LookupDtlsTransportByMidInternal(
const std::string& mid);
rtc::scoped_refptr<SctpTransportInterface> GetSctpTransport() const override;
void AddAdaptationResource(rtc::scoped_refptr<Resource> resource) override;
bool StartRtcEventLog(std::unique_ptr<RtcEventLogOutput> output,
int64_t output_period_ms) override;
bool StartRtcEventLog(std::unique_ptr<RtcEventLogOutput> output) override;
void StopRtcEventLog() override;
void Close() override;
rtc::Thread* signaling_thread() const final {
return factory_->signaling_thread();
}
// PeerConnectionInternal implementation.
rtc::Thread* network_thread() const final {
return factory_->network_thread();
}
rtc::Thread* worker_thread() const final { return factory_->worker_thread(); }
std::string session_id() const override {
RTC_DCHECK_RUN_ON(signaling_thread());
return session_id_;
}
bool initial_offerer() const override {
RTC_DCHECK_RUN_ON(signaling_thread());
return transport_controller_ && transport_controller_->initial_offerer();
}
std::vector<
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
GetTransceiversInternal() const override {
RTC_DCHECK_RUN_ON(signaling_thread());
return transceivers_;
}
sigslot::signal1<RtpDataChannel*>& SignalRtpDataChannelCreated() override {
return data_channel_controller_.SignalRtpDataChannelCreated();
}
sigslot::signal1<SctpDataChannel*>& SignalSctpDataChannelCreated() override {
return data_channel_controller_.SignalSctpDataChannelCreated();
}
cricket::RtpDataChannel* rtp_data_channel() const override {
return data_channel_controller_.rtp_data_channel();
}
std::vector<DataChannelStats> GetDataChannelStats() const override;
absl::optional<std::string> sctp_transport_name() const override;
cricket::CandidateStatsList GetPooledCandidateStats() const override;
std::map<std::string, std::string> GetTransportNamesByMid() const override;
std::map<std::string, cricket::TransportStats> GetTransportStatsByNames(
const std::set<std::string>& transport_names) override;
Call::Stats GetCallStats() override;
bool GetLocalCertificate(
const std::string& transport_name,
rtc::scoped_refptr<rtc::RTCCertificate>* certificate) override;
std::unique_ptr<rtc::SSLCertChain> GetRemoteSSLCertChain(
const std::string& transport_name) override;
bool IceRestartPending(const std::string& content_name) const override;
bool NeedsIceRestart(const std::string& content_name) const override;
bool GetSslRole(const std::string& content_name, rtc::SSLRole* role) override;
// Functions needed by DataChannelController
void NoteDataAddedEvent() { NoteUsageEvent(UsageEvent::DATA_ADDED); }
// Returns the observer. Will crash on CHECK if the observer is removed.
PeerConnectionObserver* Observer() const;
bool IsClosed() const {
RTC_DCHECK_RUN_ON(signaling_thread());
return signaling_state_ == PeerConnectionInterface::kClosed;
}
// Get current SSL role used by SCTP's underlying transport.
bool GetSctpSslRole(rtc::SSLRole* role);
// Handler for the "channel closed" signal
void OnSctpDataChannelClosed(DataChannelInterface* channel);
// Functions made public for testing.
void ReturnHistogramVeryQuicklyForTesting() {
RTC_DCHECK_RUN_ON(signaling_thread());
return_histogram_very_quickly_ = true;
}
void RequestUsagePatternReportForTesting();
absl::optional<std::string> sctp_mid() {
RTC_DCHECK_RUN_ON(signaling_thread());
return sctp_mid_s_;
}
protected:
~PeerConnection() override;
private:
class ImplicitCreateSessionDescriptionObserver;
friend class ImplicitCreateSessionDescriptionObserver;
class SetSessionDescriptionObserverAdapter;
friend class SetSessionDescriptionObserverAdapter;
// Represents the [[LocalIceCredentialsToReplace]] internal slot in the spec.
// It makes the next CreateOffer() produce new ICE credentials even if
// RTCOfferAnswerOptions::ice_restart is false.
// https://w3c.github.io/webrtc-pc/#dfn-localufragstoreplace
// TODO(hbos): When JsepTransportController/JsepTransport supports rollback,
// move this type of logic to JsepTransportController/JsepTransport.
class LocalIceCredentialsToReplace;
struct RtpSenderInfo {
RtpSenderInfo() : first_ssrc(0) {}
RtpSenderInfo(const std::string& stream_id,
const std::string sender_id,
uint32_t ssrc)
: stream_id(stream_id), sender_id(sender_id), first_ssrc(ssrc) {}
bool operator==(const RtpSenderInfo& other) {
return this->stream_id == other.stream_id &&
this->sender_id == other.sender_id &&
this->first_ssrc == other.first_ssrc;
}
std::string stream_id;
std::string sender_id;
// An RtpSender can have many SSRCs. The first one is used as a sort of ID
// for communicating with the lower layers.
uint32_t first_ssrc;
};
// Captures partial state to be used for rollback. Applicable only in
// Unified Plan.
class TransceiverStableState {
public:
TransceiverStableState() {}
void set_newly_created();
void SetMSectionIfUnset(absl::optional<std::string> mid,
absl::optional<size_t> mline_index);
void SetRemoteStreamIdsIfUnset(const std::vector<std::string>& ids);
absl::optional<std::string> mid() const { return mid_; }
absl::optional<size_t> mline_index() const { return mline_index_; }
absl::optional<std::vector<std::string>> remote_stream_ids() const {
return remote_stream_ids_;
}
bool has_m_section() const { return has_m_section_; }
bool newly_created() const { return newly_created_; }
private:
absl::optional<std::string> mid_;
absl::optional<size_t> mline_index_;
absl::optional<std::vector<std::string>> remote_stream_ids_;
// Indicates that mid value from stable state has been captured and
// that rollback has to restore the transceiver. Also protects against
// subsequent overwrites.
bool has_m_section_ = false;
// Indicates that the transceiver was created as part of applying a
// description to track potential need for removing transceiver during
// rollback.
bool newly_created_ = false;
};
// Implements MessageHandler.
void OnMessage(rtc::Message* msg) override;
// Plan B helpers for getting the voice/video media channels for the single
// audio/video transceiver, if it exists.
cricket::VoiceMediaChannel* voice_media_channel() const
RTC_RUN_ON(signaling_thread());
cricket::VideoMediaChannel* video_media_channel() const
RTC_RUN_ON(signaling_thread());
std::vector<rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>>
GetSendersInternal() const RTC_RUN_ON(signaling_thread());
std::vector<
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>>
GetReceiversInternal() const RTC_RUN_ON(signaling_thread());
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
GetAudioTransceiver() const RTC_RUN_ON(signaling_thread());
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
GetVideoTransceiver() const RTC_RUN_ON(signaling_thread());
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
GetFirstAudioTransceiver() const RTC_RUN_ON(signaling_thread());
// Implementation of the offer/answer exchange operations. These are chained
// onto the |operations_chain_| when the public CreateOffer(), CreateAnswer(),
// SetLocalDescription() and SetRemoteDescription() methods are invoked.
void DoCreateOffer(
const RTCOfferAnswerOptions& options,
rtc::scoped_refptr<CreateSessionDescriptionObserver> observer);
void DoCreateAnswer(
const RTCOfferAnswerOptions& options,
rtc::scoped_refptr<CreateSessionDescriptionObserver> observer);
void DoSetLocalDescription(
std::unique_ptr<SessionDescriptionInterface> desc,
rtc::scoped_refptr<SetLocalDescriptionObserverInterface> observer);
void DoSetRemoteDescription(
std::unique_ptr<SessionDescriptionInterface> desc,
rtc::scoped_refptr<SetRemoteDescriptionObserverInterface> observer);
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());
int usage_event_accumulator_ RTC_GUARDED_BY(signaling_thread()) = 0;
bool return_histogram_very_quickly_ RTC_GUARDED_BY(signaling_thread()) =
false;
// This object should be used to generate any SSRC that is not explicitly
// specified by the user (or by the remote party).
// The generator is not used directly, instead it is passed on to the
// channel manager and the session description factory.
rtc::UniqueRandomIdGenerator ssrc_generator_
RTC_GUARDED_BY(signaling_thread());
// A video bitrate allocator factory.
// This can injected using the PeerConnectionDependencies,
// or else the CreateBuiltinVideoBitrateAllocatorFactory() will be called.
// Note that one can still choose to override this in a MediaEngine
// if one wants too.
std::unique_ptr<webrtc::VideoBitrateAllocatorFactory>
video_bitrate_allocator_factory_;
std::unique_ptr<LocalIceCredentialsToReplace>
local_ice_credentials_to_replace_ RTC_GUARDED_BY(signaling_thread());
bool is_negotiation_needed_ RTC_GUARDED_BY(signaling_thread()) = false;
DataChannelController data_channel_controller_;
rtc::WeakPtrFactory<PeerConnection> weak_ptr_factory_
RTC_GUARDED_BY(signaling_thread());
};
} // namespace webrtc
#endif // PC_PEER_CONNECTION_H_
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