907abe4411
Reason for revert: Reverting because it broke an RTP data channel test on the FYI bots. Original issue's description: > Modified PeerConnection and WebRtcSession for end-to-end QuicDataChannel usage. > > To allow end-to-end QuicDataChannel usage with a > PeerConnection, RTCConfiguration has been modified to > include a boolean for whether to do QUIC, since negotiation of > QUIC is not implemented. If one peer does QUIC, then it will be > assumed that the other peer must do QUIC or the connection > will fail. > > PeerConnection has been modified to create data channels of type > QuicDataChannel when the peer wants to do QUIC. > > WebRtcSession has ben modified to use a QuicDataTransport > instead of a DtlsTransportChannelWrapper/DataChannel > when QUIC should be used > > QuicDataTransport implements the generic functions of > BaseChannel to manage the QuicTransportChannel. > > Committed: https://crrev.com/34b54c36a533dadb6ceb70795119194e6f530ef5 > Cr-Commit-Position: refs/heads/master@{#13645} TBR=pthatcher@webrtc.org,zhihuang@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true Review-Url: https://codereview.webrtc.org/2206793007 Cr-Commit-Position: refs/heads/master@{#13647}
108 lines
4.4 KiB
C++
108 lines
4.4 KiB
C++
/*
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* Copyright 2013 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 WEBRTC_API_TEST_PEERCONNECTIONTESTWRAPPER_H_
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#define WEBRTC_API_TEST_PEERCONNECTIONTESTWRAPPER_H_
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#include <memory>
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#include "webrtc/api/peerconnectioninterface.h"
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#include "webrtc/api/test/fakeaudiocapturemodule.h"
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#include "webrtc/api/test/fakeconstraints.h"
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#include "webrtc/api/test/fakevideotrackrenderer.h"
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#include "webrtc/base/sigslot.h"
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class PeerConnectionTestWrapper
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: public webrtc::PeerConnectionObserver,
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public webrtc::CreateSessionDescriptionObserver,
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public sigslot::has_slots<> {
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public:
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static void Connect(PeerConnectionTestWrapper* caller,
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PeerConnectionTestWrapper* callee);
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PeerConnectionTestWrapper(const std::string& name,
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rtc::Thread* network_thread,
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rtc::Thread* worker_thread);
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virtual ~PeerConnectionTestWrapper();
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bool CreatePc(const webrtc::MediaConstraintsInterface* constraints);
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rtc::scoped_refptr<webrtc::DataChannelInterface> CreateDataChannel(
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const std::string& label,
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const webrtc::DataChannelInit& init);
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// Implements PeerConnectionObserver.
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virtual void OnSignalingChange(
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webrtc::PeerConnectionInterface::SignalingState new_state) {}
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virtual void OnStateChange(
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webrtc::PeerConnectionObserver::StateType state_changed) {}
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virtual void OnAddStream(
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rtc::scoped_refptr<webrtc::MediaStreamInterface> stream);
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virtual void OnRemoveStream(
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rtc::scoped_refptr<webrtc::MediaStreamInterface> stream) {}
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virtual void OnDataChannel(
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rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel);
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virtual void OnRenegotiationNeeded() {}
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virtual void OnIceConnectionChange(
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webrtc::PeerConnectionInterface::IceConnectionState new_state) {}
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virtual void OnIceGatheringChange(
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webrtc::PeerConnectionInterface::IceGatheringState new_state) {}
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virtual void OnIceCandidate(const webrtc::IceCandidateInterface* candidate);
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virtual void OnIceComplete() {}
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// Implements CreateSessionDescriptionObserver.
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virtual void OnSuccess(webrtc::SessionDescriptionInterface* desc);
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virtual void OnFailure(const std::string& error) {}
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void CreateOffer(const webrtc::MediaConstraintsInterface* constraints);
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void CreateAnswer(const webrtc::MediaConstraintsInterface* constraints);
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void ReceiveOfferSdp(const std::string& sdp);
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void ReceiveAnswerSdp(const std::string& sdp);
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void AddIceCandidate(const std::string& sdp_mid, int sdp_mline_index,
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const std::string& candidate);
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void WaitForCallEstablished();
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void WaitForConnection();
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void WaitForAudio();
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void WaitForVideo();
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void GetAndAddUserMedia(
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bool audio, const webrtc::FakeConstraints& audio_constraints,
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bool video, const webrtc::FakeConstraints& video_constraints);
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// sigslots
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sigslot::signal1<std::string*> SignalOnIceCandidateCreated;
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sigslot::signal3<const std::string&,
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int,
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const std::string&> SignalOnIceCandidateReady;
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sigslot::signal1<std::string*> SignalOnSdpCreated;
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sigslot::signal1<const std::string&> SignalOnSdpReady;
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sigslot::signal1<webrtc::DataChannelInterface*> SignalOnDataChannel;
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private:
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void SetLocalDescription(const std::string& type, const std::string& sdp);
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void SetRemoteDescription(const std::string& type, const std::string& sdp);
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bool CheckForConnection();
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bool CheckForAudio();
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bool CheckForVideo();
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rtc::scoped_refptr<webrtc::MediaStreamInterface> GetUserMedia(
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bool audio, const webrtc::FakeConstraints& audio_constraints,
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bool video, const webrtc::FakeConstraints& video_constraints);
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std::string name_;
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rtc::Thread* const network_thread_;
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rtc::Thread* const worker_thread_;
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rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_;
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rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface>
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peer_connection_factory_;
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rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;
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std::unique_ptr<webrtc::FakeVideoTrackRenderer> renderer_;
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};
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#endif // WEBRTC_API_TEST_PEERCONNECTIONTESTWRAPPER_H_
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