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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,
(...skipping 10 matching lines...) Expand all
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"
32 #include "talk/app/webrtc/mediastream.h"
33 #include "talk/app/webrtc/remoteaudiosource.h"
34 #include "talk/app/webrtc/rtpreceiver.h"
35 #include "talk/app/webrtc/rtpsender.h"
36 #include "talk/app/webrtc/streamcollection.h"
37 #include "talk/app/webrtc/videosource.h"
38 #include "talk/app/webrtc/videotrack.h"
39 #include "testing/gmock/include/gmock/gmock.h" 31 #include "testing/gmock/include/gmock/gmock.h"
40 #include "testing/gtest/include/gtest/gtest.h" 32 #include "testing/gtest/include/gtest/gtest.h"
33 #include "webrtc/api/audiotrack.h"
34 #include "webrtc/api/mediastream.h"
35 #include "webrtc/api/remoteaudiosource.h"
36 #include "webrtc/api/rtpreceiver.h"
37 #include "webrtc/api/rtpsender.h"
38 #include "webrtc/api/streamcollection.h"
39 #include "webrtc/api/videosource.h"
40 #include "webrtc/api/videotrack.h"
41 #include "webrtc/base/gunit.h" 41 #include "webrtc/base/gunit.h"
42 #include "webrtc/media/base/fakevideocapturer.h" 42 #include "webrtc/media/base/fakevideocapturer.h"
43 #include "webrtc/media/base/mediachannel.h" 43 #include "webrtc/media/base/mediachannel.h"
44 44
45 using ::testing::_; 45 using ::testing::_;
46 using ::testing::Exactly; 46 using ::testing::Exactly;
47 47
48 static const char kStreamLabel1[] = "local_stream_1"; 48 static const char kStreamLabel1[] = "local_stream_1";
49 static const char kVideoTrackId[] = "video_1"; 49 static const char kVideoTrackId[] = "video_1";
50 static const char kAudioTrackId[] = "audio_1"; 50 static const char kAudioTrackId[] = "audio_1";
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506 video_track_->GetSource()->GetVideoCapturer())); 506 video_track_->GetSource()->GetVideoCapturer()));
507 EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc2, true, _)); 507 EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc2, true, _));
508 sender->SetSsrc(kVideoSsrc2); 508 sender->SetSsrc(kVideoSsrc2);
509 509
510 // Calls expected from destructor. 510 // Calls expected from destructor.
511 EXPECT_CALL(video_provider_, SetCaptureDevice(kVideoSsrc2, nullptr)).Times(1); 511 EXPECT_CALL(video_provider_, SetCaptureDevice(kVideoSsrc2, nullptr)).Times(1);
512 EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc2, false, _)).Times(1); 512 EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc2, false, _)).Times(1);
513 } 513 }
514 514
515 } // namespace webrtc 515 } // namespace webrtc
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