Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(398)

Unified Diff: webrtc/api/test/peerconnectiontestwrapper.h

Issue 2514883002: Create //webrtc/api:libjingle_peerconnection_api + refactorings. (Closed)
Patch Set: Rebase Created 3 years, 11 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
« no previous file with comments | « webrtc/api/test/mockpeerconnectionobservers.h ('k') | webrtc/api/test/peerconnectiontestwrapper.cc » ('j') | no next file with comments »
Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
Index: webrtc/api/test/peerconnectiontestwrapper.h
diff --git a/webrtc/api/test/peerconnectiontestwrapper.h b/webrtc/api/test/peerconnectiontestwrapper.h
deleted file mode 100644
index 6433e8fb05546e60f65736ef0ac04bfccb6f5265..0000000000000000000000000000000000000000
--- a/webrtc/api/test/peerconnectiontestwrapper.h
+++ /dev/null
@@ -1,118 +0,0 @@
-/*
- * 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_
« no previous file with comments | « webrtc/api/test/mockpeerconnectionobservers.h ('k') | webrtc/api/test/peerconnectiontestwrapper.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698