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/*
* Copyright 2022 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_SDP_METHODS_H_
#define PC_PEER_CONNECTION_SDP_METHODS_H_
#include <map>
#include <memory>
#include <set>
#include <string>
#include <vector>
#include "api/peer_connection_interface.h"
#include "pc/jsep_transport_controller.h"
#include "pc/peer_connection_message_handler.h"
#include "pc/sctp_data_channel.h"
#include "pc/usage_pattern.h"
namespace webrtc {
class DataChannelController;
class RtpTransmissionManager;
class StatsCollector;
// 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 SdpFormatReceived for the given remote description.
virtual void ReportSdpFormatReceived(
const SessionDescriptionInterface& remote_description) = 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 StatsCollector* stats() = 0;
// Returns the observer. Will crash on CHECK if the observer is removed.
virtual PeerConnectionObserver* Observer() const = 0;
virtual bool GetSctpSslRole(rtc::SSLRole* role) = 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;
virtual absl::optional<std::string> GetDataMid() const = 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;
virtual bool SetupDataChannelTransport_n(const std::string& mid) = 0;
virtual void TeardownDataChannelTransport_n() = 0;
virtual void SetSctpDataMid(const std::string& mid) = 0;
virtual void ResetSctpDataMid() = 0;
virtual const FieldTrialsView& trials() const = 0;
};
} // namespace webrtc
#endif // PC_PEER_CONNECTION_SDP_METHODS_H_
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