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

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 2012 Google Inc. 3 * Copyright 2012 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,
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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 #include <string> 28 #include <string>
29 #include <utility> 29 #include <utility>
30 30
31 #include "talk/app/webrtc/audiotrack.h" 31 #include "talk/session/media/mediasession.h"
32 #include "talk/app/webrtc/jsepsessiondescription.h" 32 #include "webrtc/api/audiotrack.h"
33 #include "talk/app/webrtc/mediastream.h" 33 #include "webrtc/api/jsepsessiondescription.h"
34 #include "talk/app/webrtc/mediastreaminterface.h" 34 #include "webrtc/api/mediastream.h"
35 #include "talk/app/webrtc/peerconnection.h" 35 #include "webrtc/api/mediastreaminterface.h"
36 #include "talk/app/webrtc/peerconnectioninterface.h" 36 #include "webrtc/api/peerconnection.h"
37 #include "talk/app/webrtc/rtpreceiverinterface.h" 37 #include "webrtc/api/peerconnectioninterface.h"
38 #include "talk/app/webrtc/rtpsenderinterface.h" 38 #include "webrtc/api/rtpreceiverinterface.h"
39 #include "talk/app/webrtc/streamcollection.h" 39 #include "webrtc/api/rtpsenderinterface.h"
40 #include "webrtc/api/streamcollection.h"
40 #ifdef WEBRTC_ANDROID 41 #ifdef WEBRTC_ANDROID
41 #include "talk/app/webrtc/test/androidtestinitializer.h" 42 #include "webrtc/api/test/androidtestinitializer.h"
42 #endif 43 #endif
43 #include "talk/app/webrtc/test/fakeconstraints.h" 44 #include "webrtc/api/test/fakeconstraints.h"
44 #include "talk/app/webrtc/test/fakedtlsidentitystore.h" 45 #include "webrtc/api/test/fakedtlsidentitystore.h"
45 #include "talk/app/webrtc/test/mockpeerconnectionobservers.h" 46 #include "webrtc/api/test/mockpeerconnectionobservers.h"
46 #include "talk/app/webrtc/test/testsdpstrings.h" 47 #include "webrtc/api/test/testsdpstrings.h"
47 #include "talk/app/webrtc/videosource.h" 48 #include "webrtc/api/videosource.h"
48 #include "talk/app/webrtc/videotrack.h" 49 #include "webrtc/api/videotrack.h"
49 #include "talk/session/media/mediasession.h"
50 #include "webrtc/base/gunit.h" 50 #include "webrtc/base/gunit.h"
51 #include "webrtc/base/scoped_ptr.h" 51 #include "webrtc/base/scoped_ptr.h"
52 #include "webrtc/base/ssladapter.h" 52 #include "webrtc/base/ssladapter.h"
53 #include "webrtc/base/sslstreamadapter.h" 53 #include "webrtc/base/sslstreamadapter.h"
54 #include "webrtc/base/stringutils.h" 54 #include "webrtc/base/stringutils.h"
55 #include "webrtc/base/thread.h" 55 #include "webrtc/base/thread.h"
56 #include "webrtc/media/base/fakevideocapturer.h" 56 #include "webrtc/media/base/fakevideocapturer.h"
57 #include "webrtc/media/sctp/sctpdataengine.h" 57 #include "webrtc/media/sctp/sctpdataengine.h"
58 #include "webrtc/p2p/client/fakeportallocator.h" 58 #include "webrtc/p2p/client/fakeportallocator.h"
59 59
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2506 FakeConstraints updated_answer_c; 2506 FakeConstraints updated_answer_c;
2507 answer_c.SetMandatoryReceiveAudio(false); 2507 answer_c.SetMandatoryReceiveAudio(false);
2508 answer_c.SetMandatoryReceiveVideo(false); 2508 answer_c.SetMandatoryReceiveVideo(false);
2509 2509
2510 cricket::MediaSessionOptions updated_answer_options; 2510 cricket::MediaSessionOptions updated_answer_options;
2511 EXPECT_TRUE( 2511 EXPECT_TRUE(
2512 ParseConstraintsForAnswer(&updated_answer_c, &updated_answer_options)); 2512 ParseConstraintsForAnswer(&updated_answer_c, &updated_answer_options));
2513 EXPECT_TRUE(updated_answer_options.has_audio()); 2513 EXPECT_TRUE(updated_answer_options.has_audio());
2514 EXPECT_TRUE(updated_answer_options.has_video()); 2514 EXPECT_TRUE(updated_answer_options.has_video());
2515 } 2515 }
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