9763d56464
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 Review-Url: https://codereview.webrtc.org/2166873002 Cr-Original-Commit-Position: refs/heads/master@{#13645} Cr-Commit-Position: refs/heads/master@{#13657}
110 lines
4.5 KiB
C++
110 lines
4.5 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(
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const webrtc::MediaConstraintsInterface* constraints,
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const webrtc::PeerConnectionInterface::RTCConfiguration& config);
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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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