From 6e70aa5905b4b2b4d609c81a0bdc048116a649c1 Mon Sep 17 00:00:00 2001 From: Tommi Date: Sun, 19 Mar 2023 11:16:23 +0100 Subject: [PATCH] Delete unused peer_connection_sdp_methods target Bug: none Change-Id: Id911670035c517556648cb601c122798544f4b58 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298303 Reviewed-by: Mirko Bonadei Commit-Queue: Tomas Gunnarsson Reviewed-by: Harald Alvestrand Cr-Commit-Position: refs/heads/main@{#39599} --- pc/BUILD.gn | 15 ---- pc/peer_connection_sdp_methods.h | 131 ------------------------------- 2 files changed, 146 deletions(-) delete mode 100644 pc/peer_connection_sdp_methods.h diff --git a/pc/BUILD.gn b/pc/BUILD.gn index ea1909bc4f..c0c2010765 100644 --- a/pc/BUILD.gn +++ b/pc/BUILD.gn @@ -981,26 +981,12 @@ rtc_source_set("data_channel_controller") { ] } -rtc_source_set("peer_connection_sdp_methods") { - visibility = [ ":*" ] - sources = [ "peer_connection_sdp_methods.h" ] - deps = [ - ":jsep_transport_controller", - ":peer_connection_message_handler", - ":sctp_data_channel", - ":usage_pattern", - "../api:libjingle_peerconnection_api", - "../call:call_interfaces", - ] -} - rtc_source_set("peer_connection_internal") { visibility = [ ":*" ] sources = [ "peer_connection_internal.h" ] deps = [ ":jsep_transport_controller", ":peer_connection_message_handler", - ":peer_connection_sdp_methods", ":rtp_transceiver", ":rtp_transmission_manager", ":sctp_data_channel", @@ -1601,7 +1587,6 @@ rtc_library("rtp_transceiver") { ":channel", ":channel_interface", ":connection_context", - ":peer_connection_sdp_methods", ":proxy", ":rtp_media_utils", ":rtp_parameters_conversion", diff --git a/pc/peer_connection_sdp_methods.h b/pc/peer_connection_sdp_methods.h deleted file mode 100644 index 972ad9c7b4..0000000000 --- a/pc/peer_connection_sdp_methods.h +++ /dev/null @@ -1,131 +0,0 @@ -/* - * 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 -#include -#include -#include -#include - -#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 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& - bundle_groups_by_mid) = 0; - - virtual absl::optional GetDataMid() const = 0; - // Internal implementation for AddTransceiver family of methods. If - // `fire_callback` is set, fires OnRenegotiationNeeded callback if successful. - virtual RTCErrorOr> - AddTransceiver(cricket::MediaType media_type, - rtc::scoped_refptr 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_