Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(387)

Side by Side Diff: webrtc/api/peerconnectioninterface_unittest.cc

Issue 2381013002: Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/ (Closed)
Patch Set: rebase Created 4 years, 2 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
« no previous file with comments | « webrtc/api/mediastream_unittest.cc ('k') | webrtc/api/proxy_unittest.cc » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
1 /* 1 /*
2 * Copyright 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright 2012 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
11 #include <memory> 11 #include <memory>
12 #include <string> 12 #include <string>
13 #include <utility> 13 #include <utility>
14 14
15 #include "webrtc/test/gmock.h"
16 #include "webrtc/api/audiotrack.h" 15 #include "webrtc/api/audiotrack.h"
17 #include "webrtc/api/jsepsessiondescription.h" 16 #include "webrtc/api/jsepsessiondescription.h"
18 #include "webrtc/api/mediastream.h" 17 #include "webrtc/api/mediastream.h"
19 #include "webrtc/api/mediastreaminterface.h" 18 #include "webrtc/api/mediastreaminterface.h"
20 #include "webrtc/api/peerconnection.h" 19 #include "webrtc/api/peerconnection.h"
21 #include "webrtc/api/peerconnectioninterface.h" 20 #include "webrtc/api/peerconnectioninterface.h"
22 #include "webrtc/api/rtpreceiverinterface.h" 21 #include "webrtc/api/rtpreceiverinterface.h"
23 #include "webrtc/api/rtpsenderinterface.h" 22 #include "webrtc/api/rtpsenderinterface.h"
24 #include "webrtc/api/streamcollection.h" 23 #include "webrtc/api/streamcollection.h"
25 #ifdef WEBRTC_ANDROID
26 #include "webrtc/api/test/androidtestinitializer.h"
27 #endif
28 #include "webrtc/api/test/fakeconstraints.h" 24 #include "webrtc/api/test/fakeconstraints.h"
29 #include "webrtc/api/test/fakertccertificategenerator.h" 25 #include "webrtc/api/test/fakertccertificategenerator.h"
30 #include "webrtc/api/test/fakevideotracksource.h" 26 #include "webrtc/api/test/fakevideotracksource.h"
31 #include "webrtc/api/test/mockpeerconnectionobservers.h" 27 #include "webrtc/api/test/mockpeerconnectionobservers.h"
32 #include "webrtc/api/test/testsdpstrings.h" 28 #include "webrtc/api/test/testsdpstrings.h"
33 #include "webrtc/api/videocapturertracksource.h" 29 #include "webrtc/api/videocapturertracksource.h"
34 #include "webrtc/api/videotrack.h" 30 #include "webrtc/api/videotrack.h"
35 #include "webrtc/base/gunit.h" 31 #include "webrtc/base/gunit.h"
36 #include "webrtc/base/ssladapter.h" 32 #include "webrtc/base/ssladapter.h"
37 #include "webrtc/base/sslstreamadapter.h" 33 #include "webrtc/base/sslstreamadapter.h"
38 #include "webrtc/base/stringutils.h" 34 #include "webrtc/base/stringutils.h"
39 #include "webrtc/base/thread.h" 35 #include "webrtc/base/thread.h"
40 #include "webrtc/media/base/fakevideocapturer.h" 36 #include "webrtc/media/base/fakevideocapturer.h"
41 #include "webrtc/media/sctp/sctpdataengine.h" 37 #include "webrtc/media/sctp/sctpdataengine.h"
42 #include "webrtc/p2p/base/fakeportallocator.h" 38 #include "webrtc/p2p/base/fakeportallocator.h"
43 #include "webrtc/p2p/base/faketransportcontroller.h" 39 #include "webrtc/p2p/base/faketransportcontroller.h"
44 #include "webrtc/pc/mediasession.h" 40 #include "webrtc/pc/mediasession.h"
41 #include "webrtc/test/gmock.h"
42
43 #ifdef WEBRTC_ANDROID
44 #include "webrtc/api/test/androidtestinitializer.h"
45 #endif
45 46
46 static const char kStreamLabel1[] = "local_stream_1"; 47 static const char kStreamLabel1[] = "local_stream_1";
47 static const char kStreamLabel2[] = "local_stream_2"; 48 static const char kStreamLabel2[] = "local_stream_2";
48 static const char kStreamLabel3[] = "local_stream_3"; 49 static const char kStreamLabel3[] = "local_stream_3";
49 static const int kDefaultStunPort = 3478; 50 static const int kDefaultStunPort = 3478;
50 static const char kStunAddressOnly[] = "stun:address"; 51 static const char kStunAddressOnly[] = "stun:address";
51 static const char kStunInvalidPort[] = "stun:address:-1"; 52 static const char kStunInvalidPort[] = "stun:address:-1";
52 static const char kStunAddressPortAndMore1[] = "stun:address:port:more"; 53 static const char kStunAddressPortAndMore1[] = "stun:address:port:more";
53 static const char kStunAddressPortAndMore2[] = "stun:address:port more"; 54 static const char kStunAddressPortAndMore2[] = "stun:address:port more";
54 static const char kTurnIceServerUri[] = "turn:user@turn.example.org"; 55 static const char kTurnIceServerUri[] = "turn:user@turn.example.org";
(...skipping 2835 matching lines...) Expand 10 before | Expand all | Expand 10 after
2890 FakeConstraints updated_answer_c; 2891 FakeConstraints updated_answer_c;
2891 answer_c.SetMandatoryReceiveAudio(false); 2892 answer_c.SetMandatoryReceiveAudio(false);
2892 answer_c.SetMandatoryReceiveVideo(false); 2893 answer_c.SetMandatoryReceiveVideo(false);
2893 2894
2894 cricket::MediaSessionOptions updated_answer_options; 2895 cricket::MediaSessionOptions updated_answer_options;
2895 EXPECT_TRUE( 2896 EXPECT_TRUE(
2896 ParseConstraintsForAnswer(&updated_answer_c, &updated_answer_options)); 2897 ParseConstraintsForAnswer(&updated_answer_c, &updated_answer_options));
2897 EXPECT_TRUE(updated_answer_options.has_audio()); 2898 EXPECT_TRUE(updated_answer_options.has_audio());
2898 EXPECT_TRUE(updated_answer_options.has_video()); 2899 EXPECT_TRUE(updated_answer_options.has_video());
2899 } 2900 }
OLDNEW
« no previous file with comments | « webrtc/api/mediastream_unittest.cc ('k') | webrtc/api/proxy_unittest.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698