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3a034e15b4
Done in preparation for some threading changes that would be quite messy if implemented with the class as-is. This results in some code duplication, but is preferable to one class having two completely different modes of operation. RTP data channels are in the process of being removed anyway, so the duplicated code won't last forever. Bug: webrtc:9883 Change-Id: Idfd41a669b56a4bb4819572e4a264a4ffaaba9c0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178940 Commit-Queue: Taylor <deadbeef@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31691}
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_PEER_CONNECTION_INTERNAL_H_
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#define PC_PEER_CONNECTION_INTERNAL_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 "call/call.h"
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#include "pc/rtp_data_channel.h"
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#include "pc/rtp_transceiver.h"
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#include "pc/sctp_data_channel.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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virtual sigslot::signal1<RtpDataChannel*>& SignalRtpDataChannelCreated() = 0;
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virtual sigslot::signal1<SctpDataChannel*>&
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SignalSctpDataChannelCreated() = 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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// Call on the network thread to fetch stats for all the data channels.
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// TODO(tommi): Make pure virtual after downstream updates.
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virtual std::vector<DataChannelStats> GetDataChannelStats() const {
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return {};
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}
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virtual absl::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_PEER_CONNECTION_INTERNAL_H_
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