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authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 19:33:14 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 19:33:14 +0000
commit36d22d82aa202bb199967e9512281e9a53db42c9 (patch)
tree105e8c98ddea1c1e4784a60a5a6410fa416be2de /third_party/libwebrtc/pc/peer_connection_internal.h
parentInitial commit. (diff)
downloadfirefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.tar.xz
firefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.zip
Adding upstream version 115.7.0esr.upstream/115.7.0esr
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
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+/*
+ * 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 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;
+
+ 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 sigslot::has_slots<> {
+ 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;
+
+ virtual sigslot::signal1<SctpDataChannel*>&
+ SignalSctpDataChannelCreated() = 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 the "channel closed" signal
+ virtual void OnSctpDataChannelClosed(DataChannelInterface* channel) {}
+};
+
+} // namespace webrtc
+
+#endif // PC_PEER_CONNECTION_INTERNAL_H_