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 <mbonadei@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39599}
This commit is contained in:
Tommi
WebRTC LUCI CQ
parent
1c6b3a3fc5
commit
6e70aa5905
15
pc/BUILD.gn
15
pc/BUILD.gn
@ -981,26 +981,12 @@ rtc_source_set("data_channel_controller") {
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]
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}
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rtc_source_set("peer_connection_sdp_methods") {
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visibility = [ ":*" ]
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sources = [ "peer_connection_sdp_methods.h" ]
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deps = [
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":jsep_transport_controller",
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":peer_connection_message_handler",
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":sctp_data_channel",
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":usage_pattern",
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"../api:libjingle_peerconnection_api",
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"../call:call_interfaces",
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]
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}
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rtc_source_set("peer_connection_internal") {
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visibility = [ ":*" ]
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sources = [ "peer_connection_internal.h" ]
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deps = [
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":jsep_transport_controller",
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":peer_connection_message_handler",
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":peer_connection_sdp_methods",
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":rtp_transceiver",
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":rtp_transmission_manager",
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":sctp_data_channel",
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@ -1601,7 +1587,6 @@ rtc_library("rtp_transceiver") {
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":channel",
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":channel_interface",
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":connection_context",
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":peer_connection_sdp_methods",
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":proxy",
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":rtp_media_utils",
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":rtp_parameters_conversion",
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@ -1,131 +0,0 @@
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/*
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* Copyright 2022 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef PC_PEER_CONNECTION_SDP_METHODS_H_
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#define PC_PEER_CONNECTION_SDP_METHODS_H_
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#include <map>
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#include <memory>
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#include <set>
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#include <string>
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#include <vector>
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#include "api/peer_connection_interface.h"
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#include "pc/jsep_transport_controller.h"
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#include "pc/peer_connection_message_handler.h"
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#include "pc/sctp_data_channel.h"
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#include "pc/usage_pattern.h"
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namespace webrtc {
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class DataChannelController;
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class RtpTransmissionManager;
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class StatsCollector;
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// This interface defines the functions that are needed for
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// SdpOfferAnswerHandler to access PeerConnection internal state.
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class PeerConnectionSdpMethods {
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public:
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virtual ~PeerConnectionSdpMethods() = default;
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// The SDP session ID as defined by RFC 3264.
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virtual std::string session_id() const = 0;
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// Returns true if the ICE restart flag above was set, and no ICE restart has
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// occurred yet for this transport (by applying a local description with
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// changed ufrag/password). If the transport has been deleted as a result of
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// bundling, returns false.
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virtual bool NeedsIceRestart(const std::string& content_name) const = 0;
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virtual absl::optional<std::string> sctp_mid() const = 0;
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// Functions below this comment are known to only be accessed
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// from SdpOfferAnswerHandler.
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// Return a pointer to the active configuration.
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virtual const PeerConnectionInterface::RTCConfiguration* configuration()
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const = 0;
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// Report the UMA metric SdpFormatReceived for the given remote description.
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virtual void ReportSdpFormatReceived(
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const SessionDescriptionInterface& remote_description) = 0;
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// Report the UMA metric BundleUsage for the given remote description.
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virtual void ReportSdpBundleUsage(
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const SessionDescriptionInterface& remote_description) = 0;
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virtual PeerConnectionMessageHandler* message_handler() = 0;
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virtual RtpTransmissionManager* rtp_manager() = 0;
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virtual const RtpTransmissionManager* rtp_manager() const = 0;
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virtual bool dtls_enabled() const = 0;
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virtual const PeerConnectionFactoryInterface::Options* options() const = 0;
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// Returns the CryptoOptions for this PeerConnection. This will always
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// return the RTCConfiguration.crypto_options if set and will only default
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// back to the PeerConnectionFactory settings if nothing was set.
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virtual CryptoOptions GetCryptoOptions() = 0;
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virtual JsepTransportController* transport_controller_s() = 0;
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virtual JsepTransportController* transport_controller_n() = 0;
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virtual DataChannelController* data_channel_controller() = 0;
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virtual cricket::PortAllocator* port_allocator() = 0;
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virtual StatsCollector* stats() = 0;
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// Returns the observer. Will crash on CHECK if the observer is removed.
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virtual PeerConnectionObserver* Observer() const = 0;
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virtual bool GetSctpSslRole(rtc::SSLRole* role) = 0;
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virtual PeerConnectionInterface::IceConnectionState
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ice_connection_state_internal() = 0;
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virtual void SetIceConnectionState(
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PeerConnectionInterface::IceConnectionState new_state) = 0;
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virtual void NoteUsageEvent(UsageEvent event) = 0;
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virtual bool IsClosed() const = 0;
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// Returns true if the PeerConnection is configured to use Unified Plan
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// semantics for creating offers/answers and setting local/remote
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// descriptions. If this is true the RtpTransceiver API will also be available
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// to the user. If this is false, Plan B semantics are assumed.
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// TODO(bugs.webrtc.org/8530): Flip the default to be Unified Plan once
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// sufficient time has passed.
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virtual bool IsUnifiedPlan() const = 0;
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virtual bool ValidateBundleSettings(
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const cricket::SessionDescription* desc,
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const std::map<std::string, const cricket::ContentGroup*>&
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bundle_groups_by_mid) = 0;
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virtual absl::optional<std::string> GetDataMid() const = 0;
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// Internal implementation for AddTransceiver family of methods. If
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// `fire_callback` is set, fires OnRenegotiationNeeded callback if successful.
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virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
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AddTransceiver(cricket::MediaType media_type,
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rtc::scoped_refptr<MediaStreamTrackInterface> track,
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const RtpTransceiverInit& init,
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bool fire_callback = true) = 0;
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// Asynchronously calls SctpTransport::Start() on the network thread for
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// `sctp_mid()` if set. Called as part of setting the local description.
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virtual void StartSctpTransport(int local_port,
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int remote_port,
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int max_message_size) = 0;
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// Asynchronously adds a remote candidate on the network thread.
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virtual void AddRemoteCandidate(const std::string& mid,
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const cricket::Candidate& candidate) = 0;
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virtual Call* call_ptr() = 0;
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// Returns true if SRTP (either using DTLS-SRTP or SDES) is required by
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// this session.
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virtual bool SrtpRequired() const = 0;
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virtual bool SetupDataChannelTransport_n(const std::string& mid) = 0;
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virtual void TeardownDataChannelTransport_n() = 0;
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virtual void SetSctpDataMid(const std::string& mid) = 0;
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virtual void ResetSctpDataMid() = 0;
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virtual const FieldTrialsView& trials() const = 0;
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};
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} // namespace webrtc
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#endif // PC_PEER_CONNECTION_SDP_METHODS_H_
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