rhubarb-lip-sync/rhubarb/lib/webrtc-8d2248ff/webrtc/api/test/peerconnectiontestwrapper.h

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2016-06-21 20:13:05 +00:00
/*
* Copyright 2013 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_TEST_PEERCONNECTIONTESTWRAPPER_H_
#define WEBRTC_API_TEST_PEERCONNECTIONTESTWRAPPER_H_
#include <memory>
#include "webrtc/api/peerconnectioninterface.h"
#include "webrtc/api/test/fakeaudiocapturemodule.h"
#include "webrtc/api/test/fakeconstraints.h"
#include "webrtc/api/test/fakevideotrackrenderer.h"
#include "webrtc/base/sigslot.h"
class PeerConnectionTestWrapper
: public webrtc::PeerConnectionObserver,
public webrtc::CreateSessionDescriptionObserver,
public sigslot::has_slots<> {
public:
static void Connect(PeerConnectionTestWrapper* caller,
PeerConnectionTestWrapper* callee);
PeerConnectionTestWrapper(const std::string& name,
rtc::Thread* network_thread,
rtc::Thread* worker_thread);
virtual ~PeerConnectionTestWrapper();
bool CreatePc(const webrtc::MediaConstraintsInterface* constraints);
rtc::scoped_refptr<webrtc::DataChannelInterface> CreateDataChannel(
const std::string& label,
const webrtc::DataChannelInit& init);
// Implements PeerConnectionObserver.
virtual void OnSignalingChange(
webrtc::PeerConnectionInterface::SignalingState new_state) {}
virtual void OnStateChange(
webrtc::PeerConnectionObserver::StateType state_changed) {}
virtual void OnAddStream(
rtc::scoped_refptr<webrtc::MediaStreamInterface> stream);
virtual void OnRemoveStream(
rtc::scoped_refptr<webrtc::MediaStreamInterface> stream) {}
virtual void OnDataChannel(
rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel);
virtual void OnRenegotiationNeeded() {}
virtual void OnIceConnectionChange(
webrtc::PeerConnectionInterface::IceConnectionState new_state) {}
virtual void OnIceGatheringChange(
webrtc::PeerConnectionInterface::IceGatheringState new_state) {}
virtual void OnIceCandidate(const webrtc::IceCandidateInterface* candidate);
virtual void OnIceComplete() {}
// Implements CreateSessionDescriptionObserver.
virtual void OnSuccess(webrtc::SessionDescriptionInterface* desc);
virtual void OnFailure(const std::string& error) {}
void CreateOffer(const webrtc::MediaConstraintsInterface* constraints);
void CreateAnswer(const webrtc::MediaConstraintsInterface* constraints);
void ReceiveOfferSdp(const std::string& sdp);
void ReceiveAnswerSdp(const std::string& sdp);
void AddIceCandidate(const std::string& sdp_mid, int sdp_mline_index,
const std::string& candidate);
void WaitForCallEstablished();
void WaitForConnection();
void WaitForAudio();
void WaitForVideo();
void GetAndAddUserMedia(
bool audio, const webrtc::FakeConstraints& audio_constraints,
bool video, const webrtc::FakeConstraints& video_constraints);
// sigslots
sigslot::signal1<std::string*> SignalOnIceCandidateCreated;
sigslot::signal3<const std::string&,
int,
const std::string&> SignalOnIceCandidateReady;
sigslot::signal1<std::string*> SignalOnSdpCreated;
sigslot::signal1<const std::string&> SignalOnSdpReady;
sigslot::signal1<webrtc::DataChannelInterface*> SignalOnDataChannel;
private:
void SetLocalDescription(const std::string& type, const std::string& sdp);
void SetRemoteDescription(const std::string& type, const std::string& sdp);
bool CheckForConnection();
bool CheckForAudio();
bool CheckForVideo();
rtc::scoped_refptr<webrtc::MediaStreamInterface> GetUserMedia(
bool audio, const webrtc::FakeConstraints& audio_constraints,
bool video, const webrtc::FakeConstraints& video_constraints);
std::string name_;
rtc::Thread* const network_thread_;
rtc::Thread* const worker_thread_;
rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_;
rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface>
peer_connection_factory_;
rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;
std::unique_ptr<webrtc::FakeVideoTrackRenderer> renderer_;
};
#endif // WEBRTC_API_TEST_PEERCONNECTIONTESTWRAPPER_H_