c392866d86
There was an implementation, but it relied on SSLCertificate::GetChain, which was never implemented. Except in the fake certificate classes used by the stats collector tests, hence the tests were passing. Instead of implementing GetChain, we decided (in https://webrtc-review.googlesource.com/c/src/+/6500) to add methods that return a SSLCertChain directly, since it results in a somewhat cleaner object model. So this CL switches everything to use the "chain" methods, and gets rid of the obsolete methods and member variables. Bug: webrtc:8920 Change-Id: Ie9d7d53654ba859535462521b54c788adec7badf Reviewed-on: https://webrtc-review.googlesource.com/56961 Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org> Reviewed-by: Zhi Huang <zhihuang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22177}
94 lines
3.4 KiB
C++
94 lines
3.4 KiB
C++
/*
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* Copyright 2018 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_PEERCONNECTIONINTERNAL_H_
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#define PC_PEERCONNECTIONINTERNAL_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/peerconnectioninterface.h"
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#include "call/call.h"
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#include "pc/datachannel.h"
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#include "pc/rtptransceiver.h"
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namespace webrtc {
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// Internal interface for extra PeerConnection methods.
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class PeerConnectionInternal : public PeerConnectionInterface {
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public:
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virtual rtc::Thread* network_thread() const = 0;
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virtual rtc::Thread* worker_thread() const = 0;
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virtual rtc::Thread* signaling_thread() const = 0;
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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 we were the initial offerer.
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virtual bool initial_offerer() const = 0;
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virtual std::vector<
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rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
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GetTransceiversInternal() const = 0;
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// Get the id used as a media stream track's "id" field from ssrc.
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virtual bool GetLocalTrackIdBySsrc(uint32_t ssrc, std::string* track_id) = 0;
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virtual bool GetRemoteTrackIdBySsrc(uint32_t ssrc, std::string* track_id) = 0;
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virtual sigslot::signal1<DataChannel*>& SignalDataChannelCreated() = 0;
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// Only valid when using deprecated RTP data channels.
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virtual cricket::RtpDataChannel* rtp_data_channel() const = 0;
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virtual std::vector<rtc::scoped_refptr<DataChannel>> sctp_data_channels()
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const = 0;
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virtual rtc::Optional<std::string> sctp_content_name() const = 0;
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virtual rtc::Optional<std::string> sctp_transport_name() const = 0;
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virtual cricket::CandidateStatsList GetPooledCandidateStats() const = 0;
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// Returns a map from MID to transport name for all active media sections.
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virtual std::map<std::string, std::string> GetTransportNamesByMid() const = 0;
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// Returns a map from transport name to transport stats for all given
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// transport names.
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virtual std::map<std::string, cricket::TransportStats>
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GetTransportStatsByNames(const std::set<std::string>& transport_names) = 0;
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virtual Call::Stats GetCallStats() = 0;
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virtual bool GetLocalCertificate(
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const std::string& transport_name,
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rtc::scoped_refptr<rtc::RTCCertificate>* certificate) = 0;
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virtual std::unique_ptr<rtc::SSLCertChain> GetRemoteSSLCertChain(
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const std::string& transport_name) = 0;
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// Returns true if there was an ICE restart initiated by the remote offer.
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virtual bool IceRestartPending(const std::string& content_name) 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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// Get SSL role for an arbitrary m= section (handles bundling correctly).
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virtual bool GetSslRole(const std::string& content_name,
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rtc::SSLRole* role) = 0;
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};
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} // namespace webrtc
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#endif // PC_PEERCONNECTIONINTERNAL_H_
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