diff options
author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
---|---|---|
committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
commit | 6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch) | |
tree | a68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/pc/peer_connection_sdp_methods.h | |
parent | Initial commit. (diff) | |
download | thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.tar.xz thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.zip |
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/pc/peer_connection_sdp_methods.h')
-rw-r--r-- | third_party/libwebrtc/pc/peer_connection_sdp_methods.h | 131 |
1 files changed, 131 insertions, 0 deletions
diff --git a/third_party/libwebrtc/pc/peer_connection_sdp_methods.h b/third_party/libwebrtc/pc/peer_connection_sdp_methods.h new file mode 100644 index 0000000000..972ad9c7b4 --- /dev/null +++ b/third_party/libwebrtc/pc/peer_connection_sdp_methods.h @@ -0,0 +1,131 @@ +/* + * 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_ |