webrtc_m130/pc/peer_connection_internal.h
Tommi 840cf78600 Move Destroy/Create steps for DataChannelTransport to PeerConnection.
This moves steps from the sdp code for pc state over to the PC class
and slightly simplifies the contract between the two classes.
Moving forward it's easier to consolidate those steps in the PC
class with other grouped operations e.g. during teardown.

Also removing GetDataMid() method in favor of the sctp_mid() property.

Bug: none
Change-Id: I938f953099d327377abd94e6b2c9ece803d88e40
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324300
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40981}
2023-10-21 16:25:11 +00:00

193 lines
8.1 KiB
C++

/*
* Copyright 2018 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_INTERNAL_H_
#define PC_PEER_CONNECTION_INTERNAL_H_
#include <map>
#include <memory>
#include <set>
#include <string>
#include <vector>
#include "absl/types/optional.h"
#include "api/peer_connection_interface.h"
#include "call/call.h"
#include "modules/audio_device/include/audio_device.h"
#include "pc/jsep_transport_controller.h"
#include "pc/peer_connection_message_handler.h"
#include "pc/rtp_transceiver.h"
#include "pc/rtp_transmission_manager.h"
#include "pc/sctp_data_channel.h"
namespace webrtc {
class DataChannelController;
class LegacyStatsCollector;
// This interface defines the functions that are needed for
// SdpOfferAnswerHandler to access PeerConnection internal state.
class PeerConnectionSdpMethods {
public:
virtual ~PeerConnectionSdpMethods() = default;
// The SDP session ID as defined by RFC 3264.
virtual std::string session_id() const = 0;
// Returns true if the ICE restart flag above was set, and no ICE restart has
// occurred yet for this transport (by applying a local description with
// changed ufrag/password). If the transport has been deleted as a result of
// bundling, returns false.
virtual bool NeedsIceRestart(const std::string& content_name) const = 0;
virtual absl::optional<std::string> sctp_mid() const = 0;
// Functions below this comment are known to only be accessed
// from SdpOfferAnswerHandler.
// Return a pointer to the active configuration.
virtual const PeerConnectionInterface::RTCConfiguration* configuration()
const = 0;
// Report the UMA metric BundleUsage for the given remote description.
virtual void ReportSdpBundleUsage(
const SessionDescriptionInterface& remote_description) = 0;
virtual PeerConnectionMessageHandler* message_handler() = 0;
virtual RtpTransmissionManager* rtp_manager() = 0;
virtual const RtpTransmissionManager* rtp_manager() const = 0;
virtual bool dtls_enabled() const = 0;
virtual const PeerConnectionFactoryInterface::Options* options() const = 0;
// 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.
virtual CryptoOptions GetCryptoOptions() = 0;
virtual JsepTransportController* transport_controller_s() = 0;
virtual JsepTransportController* transport_controller_n() = 0;
virtual DataChannelController* data_channel_controller() = 0;
virtual cricket::PortAllocator* port_allocator() = 0;
virtual LegacyStatsCollector* legacy_stats() = 0;
// Returns the observer. Will crash on CHECK if the observer is removed.
virtual PeerConnectionObserver* Observer() const = 0;
virtual absl::optional<rtc::SSLRole> GetSctpSslRole_n() = 0;
virtual PeerConnectionInterface::IceConnectionState
ice_connection_state_internal() = 0;
virtual void SetIceConnectionState(
PeerConnectionInterface::IceConnectionState new_state) = 0;
virtual void NoteUsageEvent(UsageEvent event) = 0;
virtual bool IsClosed() const = 0;
// 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.
virtual bool IsUnifiedPlan() const = 0;
virtual bool ValidateBundleSettings(
const cricket::SessionDescription* desc,
const std::map<std::string, const cricket::ContentGroup*>&
bundle_groups_by_mid) = 0;
// Internal implementation for AddTransceiver family of methods. If
// `fire_callback` is set, fires OnRenegotiationNeeded callback if successful.
virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
AddTransceiver(cricket::MediaType media_type,
rtc::scoped_refptr<MediaStreamTrackInterface> track,
const RtpTransceiverInit& init,
bool fire_callback = true) = 0;
// Asynchronously calls SctpTransport::Start() on the network thread for
// `sctp_mid()` if set. Called as part of setting the local description.
virtual void StartSctpTransport(int local_port,
int remote_port,
int max_message_size) = 0;
// Asynchronously adds a remote candidate on the network thread.
virtual void AddRemoteCandidate(const std::string& mid,
const cricket::Candidate& candidate) = 0;
virtual Call* call_ptr() = 0;
// Returns true if SRTP (either using DTLS-SRTP or SDES) is required by
// this session.
virtual bool SrtpRequired() const = 0;
// Initializes the data channel transport for the peerconnection instance.
// This will have the effect that `sctp_mid()` and `sctp_transport_name()`
// will return a set value (even though it might be an empty string) and the
// dc transport will be initialized on the network thread.
virtual bool CreateDataChannelTransport(absl::string_view mid) = 0;
// Tears down the data channel transport state and clears the `sctp_mid()` and
// `sctp_transport_name()` properties.
virtual void DestroyDataChannelTransport(RTCError error) = 0;
virtual const FieldTrialsView& trials() const = 0;
virtual void ClearStatsCache() = 0;
};
// Functions defined in this class are called by other objects,
// but not by SdpOfferAnswerHandler.
class PeerConnectionInternal : public PeerConnectionInterface,
public PeerConnectionSdpMethods {
public:
virtual rtc::Thread* network_thread() const = 0;
virtual rtc::Thread* worker_thread() const = 0;
// Returns true if we were the initial offerer.
virtual bool initial_offerer() const = 0;
virtual std::vector<
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
GetTransceiversInternal() const = 0;
// Call on the network thread to fetch stats for all the data channels.
// TODO(tommi): Make pure virtual after downstream updates.
virtual std::vector<DataChannelStats> GetDataChannelStats() const {
return {};
}
virtual absl::optional<std::string> sctp_transport_name() const = 0;
virtual cricket::CandidateStatsList GetPooledCandidateStats() const = 0;
// Returns a map from transport name to transport stats for all given
// transport names.
// Must be called on the network thread.
virtual std::map<std::string, cricket::TransportStats>
GetTransportStatsByNames(const std::set<std::string>& transport_names) = 0;
virtual Call::Stats GetCallStats() = 0;
virtual absl::optional<AudioDeviceModule::Stats> GetAudioDeviceStats() = 0;
virtual bool GetLocalCertificate(
const std::string& transport_name,
rtc::scoped_refptr<rtc::RTCCertificate>* certificate) = 0;
virtual std::unique_ptr<rtc::SSLCertChain> GetRemoteSSLCertChain(
const std::string& transport_name) = 0;
// Returns true if there was an ICE restart initiated by the remote offer.
virtual bool IceRestartPending(const std::string& content_name) const = 0;
// Get SSL role for an arbitrary m= section (handles bundling correctly).
virtual bool GetSslRole(const std::string& content_name,
rtc::SSLRole* role) = 0;
// Functions needed by DataChannelController
virtual void NoteDataAddedEvent() {}
// Handler for sctp data channel state changes.
// The `channel_id` is the same unique identifier as used in
// `DataChannelStats::internal_id and
// `RTCDataChannelStats::data_channel_identifier`.
virtual void OnSctpDataChannelStateChanged(
int channel_id,
DataChannelInterface::DataState state) {}
};
} // namespace webrtc
#endif // PC_PEER_CONNECTION_INTERNAL_H_