Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(167)

Side by Side Diff: talk/app/webrtc/test/peerconnectiontestwrapper.h

Issue 1520963002: Removing webrtc::PortAllocatorFactoryInterface. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Trying to fix presubmit warning Created 4 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 unified diff | Download patch
OLDNEW
1 /* 1 /*
2 * libjingle 2 * libjingle
3 * Copyright 2013 Google Inc. 3 * Copyright 2013 Google Inc.
4 * 4 *
5 * Redistribution and use in source and binary forms, with or without 5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met: 6 * modification, are permitted provided that the following conditions are met:
7 * 7 *
8 * 1. Redistributions of source code must retain the above copyright notice, 8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer. 9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice, 10 * 2. Redistributions in binary form must reproduce the above copyright notice,
(...skipping 16 matching lines...) Expand all
27 27
28 #ifndef TALK_APP_WEBRTC_TEST_PEERCONNECTIONTESTWRAPPER_H_ 28 #ifndef TALK_APP_WEBRTC_TEST_PEERCONNECTIONTESTWRAPPER_H_
29 #define TALK_APP_WEBRTC_TEST_PEERCONNECTIONTESTWRAPPER_H_ 29 #define TALK_APP_WEBRTC_TEST_PEERCONNECTIONTESTWRAPPER_H_
30 30
31 #include "talk/app/webrtc/peerconnectioninterface.h" 31 #include "talk/app/webrtc/peerconnectioninterface.h"
32 #include "talk/app/webrtc/test/fakeaudiocapturemodule.h" 32 #include "talk/app/webrtc/test/fakeaudiocapturemodule.h"
33 #include "talk/app/webrtc/test/fakeconstraints.h" 33 #include "talk/app/webrtc/test/fakeconstraints.h"
34 #include "talk/app/webrtc/test/fakevideotrackrenderer.h" 34 #include "talk/app/webrtc/test/fakevideotrackrenderer.h"
35 #include "webrtc/base/sigslot.h" 35 #include "webrtc/base/sigslot.h"
36 36
37 namespace webrtc {
38 class DtlsIdentityStoreInterface;
39 class PortAllocatorFactoryInterface;
40 }
41
42 class PeerConnectionTestWrapper 37 class PeerConnectionTestWrapper
43 : public webrtc::PeerConnectionObserver, 38 : public webrtc::PeerConnectionObserver,
44 public webrtc::CreateSessionDescriptionObserver, 39 public webrtc::CreateSessionDescriptionObserver,
45 public sigslot::has_slots<> { 40 public sigslot::has_slots<> {
46 public: 41 public:
47 static void Connect(PeerConnectionTestWrapper* caller, 42 static void Connect(PeerConnectionTestWrapper* caller,
48 PeerConnectionTestWrapper* callee); 43 PeerConnectionTestWrapper* callee);
49 44
50 explicit PeerConnectionTestWrapper(const std::string& name); 45 explicit PeerConnectionTestWrapper(const std::string& name);
51 virtual ~PeerConnectionTestWrapper(); 46 virtual ~PeerConnectionTestWrapper();
(...skipping 51 matching lines...) Expand 10 before | Expand all | Expand 10 after
103 void SetLocalDescription(const std::string& type, const std::string& sdp); 98 void SetLocalDescription(const std::string& type, const std::string& sdp);
104 void SetRemoteDescription(const std::string& type, const std::string& sdp); 99 void SetRemoteDescription(const std::string& type, const std::string& sdp);
105 bool CheckForConnection(); 100 bool CheckForConnection();
106 bool CheckForAudio(); 101 bool CheckForAudio();
107 bool CheckForVideo(); 102 bool CheckForVideo();
108 rtc::scoped_refptr<webrtc::MediaStreamInterface> GetUserMedia( 103 rtc::scoped_refptr<webrtc::MediaStreamInterface> GetUserMedia(
109 bool audio, const webrtc::FakeConstraints& audio_constraints, 104 bool audio, const webrtc::FakeConstraints& audio_constraints,
110 bool video, const webrtc::FakeConstraints& video_constraints); 105 bool video, const webrtc::FakeConstraints& video_constraints);
111 106
112 std::string name_; 107 std::string name_;
113 rtc::scoped_refptr<webrtc::PortAllocatorFactoryInterface>
114 allocator_factory_;
115 rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_; 108 rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_;
116 rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> 109 rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface>
117 peer_connection_factory_; 110 peer_connection_factory_;
118 rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_; 111 rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;
119 rtc::scoped_ptr<webrtc::FakeVideoTrackRenderer> renderer_; 112 rtc::scoped_ptr<webrtc::FakeVideoTrackRenderer> renderer_;
120 }; 113 };
121 114
122 #endif // TALK_APP_WEBRTC_TEST_PEERCONNECTIONTESTWRAPPER_H_ 115 #endif // TALK_APP_WEBRTC_TEST_PEERCONNECTIONTESTWRAPPER_H_
OLDNEW
« no previous file with comments | « talk/app/webrtc/portallocatorfactory.cc ('k') | talk/app/webrtc/test/peerconnectiontestwrapper.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698