This provides an intermediate class for defining default, null implementations of all the PeerConnectionInterface/ PeerConnectionInternal methods. Specific fake PeerConnections then can inherit from this and only override the methods pertaining to the scenarios it will be used in. Bug: webrtc:8764 Change-Id: I7614303b5673747244053b54b839e58aada43d10 Reviewed-on: https://webrtc-review.googlesource.com/43245 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21752}
119 lines
4.4 KiB
C++
119 lines
4.4 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 <string>
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#include <vector>
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#include "api/peerconnectioninterface.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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// Statistics for all the transports of the session.
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// TODO(pthatcher): Think of a better name for this. We already have
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// a TransportStats in transport.h. Perhaps TransportsStats?
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struct SessionStats {
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std::map<std::string, cricket::TransportStats> transport_stats;
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};
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struct ChannelNamePair {
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ChannelNamePair(const std::string& content_name,
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const std::string& transport_name)
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: content_name(content_name), transport_name(transport_name) {}
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std::string content_name;
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std::string transport_name;
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};
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struct ChannelNamePairs {
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rtc::Optional<ChannelNamePair> voice;
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rtc::Optional<ChannelNamePair> video;
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rtc::Optional<ChannelNamePair> data;
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};
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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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// TODO(steveanton): Remove these.
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virtual cricket::VoiceChannel* voice_channel() const = 0;
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virtual cricket::VideoChannel* video_channel() const = 0;
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// Exposed for tests.
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virtual std::vector<
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rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
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GetTransceiversForTesting() 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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// Returns stats for all channels of all transports.
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// This avoids exposing the internal structures used to track them.
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// The parameterless version creates |ChannelNamePairs| from |voice_channel|,
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// |video_channel| and |voice_channel| if available - this requires it to be
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// called on the signaling thread - and invokes the other |GetStats|. The
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// other |GetStats| can be invoked on any thread; if not invoked on the
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// network thread a thread hop will happen.
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virtual std::unique_ptr<SessionStats> GetSessionStats_s() = 0;
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virtual std::unique_ptr<SessionStats> GetSessionStats(
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const ChannelNamePairs& channel_name_pairs) = 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::SSLCertificate> GetRemoteSSLCertificate(
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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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