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/*
* libjingle
* Copyright 2013 Google Inc.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are met:
*
* 1. Redistributions of source code must retain the above copyright notice,
* this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright notice,
* this list of conditions and the following disclaimer in the documentation
* and/or other materials provided with the distribution.
* 3. The name of the author may not be used to endorse or promote products
* derived from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef TALK_APP_WEBRTC_TEST_PEERCONNECTIONTESTWRAPPER_H_
#define TALK_APP_WEBRTC_TEST_PEERCONNECTIONTESTWRAPPER_H_
#include "talk/app/webrtc/peerconnectioninterface.h"
#include "talk/app/webrtc/test/fakeaudiocapturemodule.h"
#include "talk/app/webrtc/test/fakeconstraints.h"
#include "talk/app/webrtc/test/fakevideotrackrenderer.h"
#include "webrtc/base/sigslot.h"
namespace webrtc {
class DtlsIdentityStoreInterface;
class PortAllocatorFactoryInterface;
}
class PeerConnectionTestWrapper
: public webrtc::PeerConnectionObserver,
public webrtc::CreateSessionDescriptionObserver,
public sigslot::has_slots<> {
public:
static void Connect(PeerConnectionTestWrapper* caller,
PeerConnectionTestWrapper* callee);
explicit PeerConnectionTestWrapper(const std::string& name);
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(webrtc::MediaStreamInterface* stream);
virtual void OnRemoveStream(webrtc::MediaStreamInterface* stream) {}
virtual void OnDataChannel(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::scoped_refptr<webrtc::PortAllocatorFactoryInterface>
allocator_factory_;
rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_;
rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface>
peer_connection_factory_;
rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;
rtc::scoped_ptr<webrtc::FakeVideoTrackRenderer> renderer_;
};
#endif // TALK_APP_WEBRTC_TEST_PEERCONNECTIONTESTWRAPPER_H_
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