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webrtc/pc/peerconnection.h
Steve Anton bb19276a32 Use AsyncInvoker in PeerConnection instead of MessageHandler 
Bug: webrtc:9702
Change-Id: I89d66d1165a096601aed37b8febad60620073899
Reviewed-on: https://webrtc-review.googlesource.com/97180
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24515}
2018-08-31 20:10:30 +00:00

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/*
* 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_PEERCONNECTION_H_
#define PC_PEERCONNECTION_H_
#include <map>
#include <memory>
#include <set>
#include <string>
#include <vector>
#include "api/peerconnectioninterface.h"
#include "api/turncustomizer.h"
#include "pc/iceserverparsing.h"
#include "pc/jseptransportcontroller.h"
#include "pc/peerconnectionfactory.h"
#include "pc/peerconnectioninternal.h"
#include "pc/rtcstatscollector.h"
#include "pc/rtptransceiver.h"
#include "pc/statscollector.h"
#include "pc/streamcollection.h"
#include "pc/webrtcsessiondescriptionfactory.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 DataChannelProviderInterface,
public JsepTransportController::Observer,
public sigslot::has_slots<> {
public:
enum class UsageEvent : int {
TURN_SERVER_ADDED = 0x01,
STUN_SERVER_ADDED = 0x02,
DATA_ADDED = 0x04,
AUDIO_ADDED = 0x08,
VIDEO_ADDED = 0x10,
SET_LOCAL_DESCRIPTION_CALLED = 0x20,
SET_REMOTE_DESCRIPTION_CALLED = 0x40,
CANDIDATE_COLLECTED = 0x80,
REMOTE_CANDIDATE_ADDED = 0x100,
ICE_STATE_CONNECTED = 0x200,
CLOSE_CALLED = 0x400,
PRIVATE_CANDIDATE_COLLECTED = 0x800,
MAX_VALUE = 0x1000,
};
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;
IceGatheringState ice_gathering_state() 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;
// JSEP01
void CreateOffer(CreateSessionDescriptionObserver* observer,
const RTCOfferAnswerOptions& options) override;
void CreateAnswer(CreateSessionDescriptionObserver* observer,
const RTCOfferAnswerOptions& options) override;
void SetLocalDescription(SetSessionDescriptionObserver* observer,
SessionDescriptionInterface* desc) override;
void SetRemoteDescription(SetSessionDescriptionObserver* observer,
SessionDescriptionInterface* desc) override;
void SetRemoteDescription(
std::unique_ptr<SessionDescriptionInterface> desc,
rtc::scoped_refptr<SetRemoteDescriptionObserverInterface> observer)
override;
PeerConnectionInterface::RTCConfiguration GetConfiguration() override;
bool SetConfiguration(
const PeerConnectionInterface::RTCConfiguration& configuration,
RTCError* error) override;
bool SetConfiguration(
const PeerConnectionInterface::RTCConfiguration& configuration) override {
return SetConfiguration(configuration, nullptr);
}
bool AddIceCandidate(const IceCandidateInterface* candidate) override;
bool RemoveIceCandidates(
const std::vector<cricket::Candidate>& candidates) override;
RTCError SetBitrate(const BitrateSettings& bitrate) override;
void SetBitrateAllocationStrategy(
std::unique_ptr<rtc::BitrateAllocationStrategy>
bitrate_allocation_strategy) override;
void SetAudioPlayout(bool playout) override;
void SetAudioRecording(bool recording) override;
RTC_DEPRECATED bool StartRtcEventLog(rtc::PlatformFile file,
int64_t max_size_bytes) override;
bool StartRtcEventLog(std::unique_ptr<RtcEventLogOutput> output,
int64_t output_period_ms) override;
void StopRtcEventLog() override;
void Close() override;
// PeerConnectionInternal implementation.
rtc::Thread* network_thread() const override {
return factory_->network_thread();
}
rtc::Thread* worker_thread() const override {
return factory_->worker_thread();
}
rtc::Thread* signaling_thread() const override {
return factory_->signaling_thread();
}
std::string session_id() const override { return session_id_; }
bool initial_offerer() const override {
return transport_controller_ && transport_controller_->initial_offerer();
}
std::vector<
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
GetTransceiversInternal() const override {
return transceivers_;
}
bool GetLocalTrackIdBySsrc(uint32_t ssrc, std::string* track_id) override;
bool GetRemoteTrackIdBySsrc(uint32_t ssrc, std::string* track_id) override;
sigslot::signal1<DataChannel*>& SignalDataChannelCreated() override {
return SignalDataChannelCreated_;
}
cricket::RtpDataChannel* rtp_data_channel() const override {
return rtp_data_channel_;
}
std::vector<rtc::scoped_refptr<DataChannel>> sctp_data_channels()
const override {
return sctp_data_channels_;
}
absl::optional<std::string> sctp_content_name() const override {
return sctp_mid_;
}
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;
void ReturnHistogramVeryQuicklyForTesting() {
return_histogram_very_quickly_ = true;
}
void RequestUsagePatternReportForTesting();
protected:
~PeerConnection() override;
private:
class SetRemoteDescriptionObserverAdapter;
friend class SetRemoteDescriptionObserverAdapter;
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;
};
// 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;
cricket::VideoMediaChannel* video_media_channel() const;
std::vector<rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>>
GetSendersInternal() const;
std::vector<
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>>
GetReceiversInternal() const;
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
GetAudioTransceiver() const;
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
GetVideoTransceiver() const;
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
GetFirstAudioTransceiver() const;
void CreateAudioReceiver(MediaStreamInterface* stream,
const RtpSenderInfo& remote_sender_info);
void CreateVideoReceiver(MediaStreamInterface* stream,
const RtpSenderInfo& remote_sender_info);
rtc::scoped_refptr<RtpReceiverInterface> RemoveAndStopReceiver(
const RtpSenderInfo& remote_sender_info);
// 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);
void RemoveAudioTrack(AudioTrackInterface* track,
MediaStreamInterface* stream);
void AddVideoTrack(VideoTrackInterface* track, MediaStreamInterface* stream);
void RemoveVideoTrack(VideoTrackInterface* track,
MediaStreamInterface* stream);
// 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);
// 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);
// 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::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
FindTransceiverBySender(rtc::scoped_refptr<RtpSenderInterface> sender);
// 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::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);
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);
void SetIceConnectionState(IceConnectionState new_state);
// Called any time the IceGatheringState changes
void OnIceGatheringChange(IceGatheringState new_state);
// New ICE candidate has been gathered.
void OnIceCandidate(std::unique_ptr<IceCandidateInterface> candidate);
// Some local ICE candidates have been removed.
void OnIceCandidatesRemoved(
const std::vector<cricket::Candidate>& candidates);
// Update the state, signaling if necessary.
void ChangeSignalingState(SignalingState signaling_state);
// Signals from MediaStreamObserver.
void OnAudioTrackAdded(AudioTrackInterface* track,
MediaStreamInterface* stream);
void OnAudioTrackRemoved(AudioTrackInterface* track,
MediaStreamInterface* stream);
void OnVideoTrackAdded(VideoTrackInterface* track,
MediaStreamInterface* stream);
void OnVideoTrackRemoved(VideoTrackInterface* track,
MediaStreamInterface* stream);
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);
// 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);
// 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);
// 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);
// Returns the RtpTransceiver, if found, that is associated to the given MID.
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
GetAssociatedTransceiver(const std::string& mid) const;
// 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;
// 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;
// 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;
// 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);
void OnNegotiationNeeded();
bool IsClosed() const {
return signaling_state_ == PeerConnectionInterface::kClosed;
}
// 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);
void GetOptionsForPlanBOffer(
const PeerConnectionInterface::RTCOfferAnswerOptions&
offer_answer_options,
cricket::MediaSessionOptions* session_options);
void GetOptionsForUnifiedPlanOffer(
const PeerConnectionInterface::RTCOfferAnswerOptions&
offer_answer_options,
cricket::MediaSessionOptions* session_options);
RTCError HandleLegacyOfferOptions(const RTCOfferAnswerOptions& options);
void RemoveRecvDirectionFromReceivingTransceiversOfType(
cricket::MediaType media_type);
void AddUpToOneReceivingTransceiverOfType(cricket::MediaType media_type);
std::vector<
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
GetReceivingTransceiversOfType(cricket::MediaType media_type);
// 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);
void GetOptionsForPlanBAnswer(
const PeerConnectionInterface::RTCOfferAnswerOptions&
offer_answer_options,
cricket::MediaSessionOptions* session_options);
void GetOptionsForUnifiedPlanAnswer(
const PeerConnectionInterface::RTCOfferAnswerOptions&
offer_answer_options,
cricket::MediaSessionOptions* session_options);
// 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);
// Generates the active MediaDescriptionOptions for the local data channel
// given the specified MID.
cricket::MediaDescriptionOptions GetMediaDescriptionOptionsForActiveData(
const std::string& mid) const;
// Generates the rejected MediaDescriptionOptions for the local data channel
// given the specified MID.
cricket::MediaDescriptionOptions GetMediaDescriptionOptionsForRejectedData(
const std::string& mid) const;
// 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;
// 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);
// 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);
// 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);
// 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);
// Finds remote MediaStreams without any tracks and removes them from
// |remote_streams_| and notifies the observer that the MediaStreams no longer
// exist.
void UpdateEndedRemoteMediaStreams();
// 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);
// 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);
// 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);
void UpdateLocalRtpDataChannels(const cricket::StreamParamsVec& streams);
void UpdateRemoteRtpDataChannels(const cricket::StreamParamsVec& streams);
void UpdateClosingRtpDataChannels(
const std::vector<std::string>& active_channels,
bool is_local_update);
void CreateRemoteRtpDataChannel(const std::string& label,
uint32_t remote_ssrc);
// Creates channel and adds it to the collection of DataChannels that will
// be offered in a SessionDescription.
rtc::scoped_refptr<DataChannel> InternalCreateDataChannel(
const std::string& label,
const InternalDataChannelInit* config);
// Checks if any data channel has been added.
bool HasDataChannels() const;
void AllocateSctpSids(rtc::SSLRole role);
void OnSctpDataChannelClosed(DataChannel* channel);
void OnDataChannelDestroyed();
// Called when a valid data channel OPEN message is received.
void OnDataChannelOpenMessage(const std::string& label,
const InternalDataChannelInit& config);
// 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 {
return configuration_.sdp_semantics == SdpSemantics::kUnifiedPlan;
}
// Is there an RtpSender of the given type?
bool HasRtpSender(cricket::MediaType type) const;
// Return the RtpSender with the given track attached.
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>
FindSenderForTrack(MediaStreamTrackInterface* track) const;
// Return the RtpSender with the given id, or null if none exists.
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>
FindSenderById(const std::string& sender_id) const;
// Return the RtpReceiver with the given id, or null if none exists.
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
FindReceiverById(const std::string& receiver_id) const;
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.
DataChannel* FindDataChannelBySid(int sid) const;
// Called when first configuring the port allocator.
bool 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,
bool prune_turn_ports,
webrtc::TurnCustomizer* turn_customizer,
absl::optional<int> stun_candidate_keepalive_interval);
// 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 { return session_error_; }
const std::string& session_error_desc() const { return session_error_desc_; }
cricket::BaseChannel* GetChannel(const std::string& content_name);
// Get current SSL role used by SCTP's underlying transport.
bool GetSctpSslRole(rtc::SSLRole* role);
cricket::IceConfig ParseIceConfig(
const PeerConnectionInterface::RTCConfiguration& config) const;
// Implements DataChannelProviderInterface.
bool SendData(const cricket::SendDataParams& params,
const rtc::CopyOnWriteBuffer& payload,
cricket::SendDataResult* result) override;
bool ConnectDataChannel(DataChannel* webrtc_data_channel) override;
void DisconnectDataChannel(DataChannel* webrtc_data_channel) override;
void AddSctpDataStream(int sid) override;
void RemoveSctpDataStream(int sid) override;
bool ReadyToSendData() const override;
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() {
return pending_local_description_ ? pending_local_description_.get()
: current_local_description_.get();
}
SessionDescriptionInterface* mutable_remote_description() {
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);
bool PushdownSctpParameters_n(cricket::ContentSource source);
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();
// Destroys all BaseChannels and destroys the SCTP data channel, if present.
void DestroyAllChannels();
// 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);
// Uses all remote candidates in |remote_desc| in this session.
bool UseCandidatesInSessionDescription(
const SessionDescriptionInterface* remote_desc);
// Uses |candidate| in this session.
bool UseCandidate(const IceCandidateInterface* candidate);
// 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);
// 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);
// 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;
// Helper methods to create media channels.
cricket::VoiceChannel* CreateVoiceChannel(const std::string& mid);
cricket::VideoChannel* CreateVideoChannel(const std::string& mid);
bool CreateDataChannel(const std::string& mid);
bool CreateSctpTransport_n(const std::string& mid);
// For bundling.
void DestroySctpTransport_n();
// SctpTransport signal handlers. Needed to marshal signals from the network
// to signaling thread.
void OnSctpTransportReadyToSendData_n();
// This may be called with "false" if the direction of the m= section causes
// us to tear down the SCTP connection.
void OnSctpTransportReadyToSendData_s(bool ready);
void OnSctpTransportDataReceived_n(const cricket::ReceiveDataParams& params,
const rtc::CopyOnWriteBuffer& payload);
// Beyond just firing the signal to the signaling thread, listens to SCTP
// CONTROL messages on unused SIDs and processes them as OPEN messages.
void OnSctpTransportDataReceived_s(const cricket::ReceiveDataParams& params,
const rtc::CopyOnWriteBuffer& payload);
void OnSctpClosingProcedureStartedRemotely_n(int sid);
void OnSctpClosingProcedureComplete_n(int sid);
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);
// 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);
// Returns true if SRTP (either using DTLS-SRTP or SDES) is required by
// this session.
bool SrtpRequired() const;
// JsepTransportController signal handlers.
void OnTransportControllerConnectionState(cricket::IceConnectionState state);
void OnTransportControllerGatheringState(cricket::IceGatheringState state);
void OnTransportControllerCandidatesGathered(
const std::string& transport_name,
const std::vector<cricket::Candidate>& candidates);
void OnTransportControllerCandidatesRemoved(
const std::vector<cricket::Candidate>& candidates);
void OnTransportControllerDtlsHandshakeError(rtc::SSLHandshakeError error);
const char* SessionErrorToString(SessionError error) const;
std::string GetSessionErrorMsg();
// 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();
// 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);
void NoteUsageEvent(UsageEvent event);
void ReportUsagePattern() const;
void OnSentPacket_w(const rtc::SentPacket& sent_packet);
const std::string GetTransportName(const std::string& content_name);
// 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 and/or the SCTP data channel and clears it.
void DestroyDataChannel();
// Destroys the given BaseChannel. The channel cannot be accessed after this
// method is called.
void DestroyBaseChannel(cricket::BaseChannel* 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,
cricket::DtlsTransportInternal* dtls_transport) override;
// Returns the observer. Will crash on CHECK if the observer is removed.
PeerConnectionObserver* Observer() const;
sigslot::signal1<DataChannel*> SignalDataChannelCreated_;
// 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.
rtc::scoped_refptr<PeerConnectionFactory> factory_;
PeerConnectionObserver* observer_ = nullptr;
// The EventLog needs to outlive |call_| (and any other object that uses it).
std::unique_ptr<RtcEventLog> event_log_;
SignalingState signaling_state_ = kStable;
IceConnectionState ice_connection_state_ = kIceConnectionNew;
IceGatheringState ice_gathering_state_ = kIceGatheringNew;
PeerConnectionInterface::RTCConfiguration configuration_;
// 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_;
std::unique_ptr<cricket::PortAllocator> port_allocator_;
std::unique_ptr<rtc::SSLCertificateVerifier> tls_cert_verifier_;
int port_allocator_flags_ = 0;
// One PeerConnection has only one RTCP CNAME.
// https://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-26#section-4.9
std::string rtcp_cname_;
// Streams added via AddStream.
rtc::scoped_refptr<StreamCollection> local_streams_;
// Streams created as a result of SetRemoteDescription.
rtc::scoped_refptr<StreamCollection> remote_streams_;
std::vector<std::unique_ptr<MediaStreamObserver>> stream_observers_;
// These lists store sender info seen in local/remote descriptions.
std::vector<RtpSenderInfo> remote_audio_sender_infos_;
std::vector<RtpSenderInfo> remote_video_sender_infos_;
std::vector<RtpSenderInfo> local_audio_sender_infos_;
std::vector<RtpSenderInfo> local_video_sender_infos_;
SctpSidAllocator sid_allocator_;
// label -> DataChannel
std::map<std::string, rtc::scoped_refptr<DataChannel>> rtp_data_channels_;
std::vector<rtc::scoped_refptr<DataChannel>> sctp_data_channels_;
std::vector<rtc::scoped_refptr<DataChannel>> sctp_data_channels_to_free_;
bool remote_peer_supports_msid_ = false;
std::unique_ptr<Call> call_;
std::unique_ptr<StatsCollector> stats_; // A pointer is passed to senders_
rtc::scoped_refptr<RTCStatsCollector> stats_collector_;
std::vector<
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
transceivers_;
// MIDs that have been seen either by SetLocalDescription or
// SetRemoteDescription over the life of the PeerConnection.
std::set<std::string> seen_mids_;
SessionError session_error_ = SessionError::kNone;
std::string session_error_desc_;
std::string session_id_;
std::unique_ptr<JsepTransportController> transport_controller_;
std::unique_ptr<cricket::SctpTransportInternalFactory> sctp_factory_;
// |rtp_data_channel_| is used if in RTP data channel mode, |sctp_transport_|
// when using SCTP.
cricket::RtpDataChannel* rtp_data_channel_ = nullptr;
std::unique_ptr<cricket::SctpTransportInternal> sctp_transport_;
// |sctp_mid_| is the content name (MID) in SDP.
absl::optional<std::string> sctp_mid_;
// Value cached on signaling thread. Only updated when SctpReadyToSendData
// fires on the signaling thread.
bool sctp_ready_to_send_data_ = false;
// Same as signals provided by SctpTransport, but these are guaranteed to
// fire on the signaling thread, whereas SctpTransport fires on the networking
// thread.
// |sctp_invoker_| is used so that any signals queued on the signaling thread
// from the network thread are immediately discarded if the SctpTransport is
// destroyed (due to m= section being rejected).
// TODO(deadbeef): Use a proxy object to ensure that method calls/signals
// are marshalled to the right thread. Could almost use proxy.h for this,
// but it doesn't have a mechanism for marshalling sigslot::signals
std::unique_ptr<rtc::AsyncInvoker> sctp_invoker_;
sigslot::signal1<bool> SignalSctpReadyToSendData;
sigslot::signal2<const cricket::ReceiveDataParams&,
const rtc::CopyOnWriteBuffer&>
SignalSctpDataReceived;
sigslot::signal1<int> SignalSctpClosingProcedureStartedRemotely;
sigslot::signal1<int> SignalSctpClosingProcedureComplete;
std::unique_ptr<SessionDescriptionInterface> current_local_description_;
std::unique_ptr<SessionDescriptionInterface> pending_local_description_;
std::unique_ptr<SessionDescriptionInterface> current_remote_description_;
std::unique_ptr<SessionDescriptionInterface> pending_remote_description_;
bool dtls_enabled_ = false;
// Specifies which kind of data channel is allowed. This is controlled
// by the chrome command-line flag and constraints:
// 1. If chrome command-line switch 'enable-sctp-data-channels' is enabled,
// constraint kEnableDtlsSrtp is true, and constaint kEnableRtpDataChannels is
// not set or false, SCTP is allowed (DCT_SCTP);
// 2. If constraint kEnableRtpDataChannels is true, RTP is allowed (DCT_RTP);
// 3. If both 1&2 are false, data channel is not allowed (DCT_NONE).
cricket::DataChannelType data_channel_type_ = cricket::DCT_NONE;
// List of content names for which the remote side triggered an ICE restart.
std::set<std::string> pending_ice_restarts_;
std::unique_ptr<WebRtcSessionDescriptionFactory> webrtc_session_desc_factory_;
// Member variables for caching global options.
cricket::AudioOptions audio_options_;
cricket::VideoOptions video_options_;
int usage_event_accumulator_ = 0;
bool return_histogram_very_quickly_ = false;
rtc::AsyncInvoker async_invoker_;
};
} // namespace webrtc
#endif // PC_PEERCONNECTION_H_
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