db346a7cbe
This is an integration test using peerconnectiontestwrapper.h to set up and end to end test using a real PeerConnection implementation. These tests will complement rtcstatscollector_unittest.cc which collects all stats using mocks. The integration test is set up so that all stats types are returned by GetStats and verifies that expected dictionary members are defined. The test could in the future be updated to include sanity checks for the values of members. There is a sanity check that references to other stats dictionaries yield existing stats of the appropriate type, but other than that members are only tested for if they are defined not. StatsCallback of rtcstatscollector_unittest.cc is moved so that it can be reused and renamed to RTCStatsObtainer. TODO: Audio stream track stats members are missing in the test. Find out if this is because of a real problem or because of testing without real devices. Do this before closing crbug.com/627816. BUG=chromium:627816 Review-Url: https://codereview.webrtc.org/2521663002 Cr-Commit-Position: refs/heads/master@{#15287}
119 lines
4.9 KiB
C++
119 lines
4.9 KiB
C++
/*
|
|
* 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:
|
|
// We need these using declarations because there are two versions of each of
|
|
// the below methods and we only override one of them.
|
|
// TODO(deadbeef): Remove once there's only one version of the methods.
|
|
using PeerConnectionObserver::OnAddStream;
|
|
using PeerConnectionObserver::OnRemoveStream;
|
|
using PeerConnectionObserver::OnDataChannel;
|
|
|
|
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,
|
|
const webrtc::PeerConnectionInterface::RTCConfiguration& config);
|
|
|
|
webrtc::PeerConnectionInterface* pc() { return peer_connection_.get(); }
|
|
|
|
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_
|