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Side by Side Diff: webrtc/api/test/peerconnectiontestwrapper.h

Issue 1610243002: Move talk/app/webrtc to webrtc/api (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Removed processing of api.gyp for Chromium builds Created 4 years, 10 months ago
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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,
11 * this list of conditions and the following disclaimer in the documentation 11 * this list of conditions and the following disclaimer in the documentation
12 * and/or other materials provided with the distribution. 12 * and/or other materials provided with the distribution.
13 * 3. The name of the author may not be used to endorse or promote products 13 * 3. The name of the author may not be used to endorse or promote products
14 * derived from this software without specific prior written permission. 14 * derived from this software without specific prior written permission.
15 * 15 *
16 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED 16 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
17 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF 17 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
18 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO 18 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
19 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, 19 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
20 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, 20 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; 21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, 22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR 23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF 24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. 25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
26 */ 26 */
27 27
28 #ifndef TALK_APP_WEBRTC_TEST_PEERCONNECTIONTESTWRAPPER_H_ 28 #ifndef WEBRTC_API_TEST_PEERCONNECTIONTESTWRAPPER_H_
29 #define TALK_APP_WEBRTC_TEST_PEERCONNECTIONTESTWRAPPER_H_ 29 #define WEBRTC_API_TEST_PEERCONNECTIONTESTWRAPPER_H_
30 30
31 #include "talk/app/webrtc/peerconnectioninterface.h" 31 #include "webrtc/api/peerconnectioninterface.h"
32 #include "talk/app/webrtc/test/fakeaudiocapturemodule.h" 32 #include "webrtc/api/test/fakeaudiocapturemodule.h"
33 #include "talk/app/webrtc/test/fakeconstraints.h" 33 #include "webrtc/api/test/fakeconstraints.h"
34 #include "talk/app/webrtc/test/fakevideotrackrenderer.h" 34 #include "webrtc/api/test/fakevideotrackrenderer.h"
35 #include "webrtc/base/sigslot.h" 35 #include "webrtc/base/sigslot.h"
36 36
37 class PeerConnectionTestWrapper 37 class PeerConnectionTestWrapper
38 : public webrtc::PeerConnectionObserver, 38 : public webrtc::PeerConnectionObserver,
39 public webrtc::CreateSessionDescriptionObserver, 39 public webrtc::CreateSessionDescriptionObserver,
40 public sigslot::has_slots<> { 40 public sigslot::has_slots<> {
41 public: 41 public:
42 static void Connect(PeerConnectionTestWrapper* caller, 42 static void Connect(PeerConnectionTestWrapper* caller,
43 PeerConnectionTestWrapper* callee); 43 PeerConnectionTestWrapper* callee);
44 44
(...skipping 60 matching lines...) Expand 10 before | Expand all | Expand 10 after
105 bool video, const webrtc::FakeConstraints& video_constraints); 105 bool video, const webrtc::FakeConstraints& video_constraints);
106 106
107 std::string name_; 107 std::string name_;
108 rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_; 108 rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_;
109 rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> 109 rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface>
110 peer_connection_factory_; 110 peer_connection_factory_;
111 rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_; 111 rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;
112 rtc::scoped_ptr<webrtc::FakeVideoTrackRenderer> renderer_; 112 rtc::scoped_ptr<webrtc::FakeVideoTrackRenderer> renderer_;
113 }; 113 };
114 114
115 #endif // TALK_APP_WEBRTC_TEST_PEERCONNECTIONTESTWRAPPER_H_ 115 #endif // WEBRTC_API_TEST_PEERCONNECTIONTESTWRAPPER_H_
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