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1 // Copyright 2014 The Chromium Authors. All rights reserved. | 1 // Copyright 2014 The Chromium Authors. All rights reserved. |
2 // Use of this source code is governed by a BSD-style license that can be | 2 // Use of this source code is governed by a BSD-style license that can be |
3 // found in the LICENSE file. | 3 // found in the LICENSE file. |
4 | 4 |
5 #include <stddef.h> | 5 #include <stddef.h> |
6 | 6 |
7 #include "base/bind.h" | 7 #include "base/bind.h" |
8 #include "base/message_loop/message_loop.h" | 8 #include "base/message_loop/message_loop.h" |
9 #include "base/run_loop.h" | 9 #include "base/run_loop.h" |
10 #include "base/threading/thread.h" | 10 #include "base/threading/thread.h" |
11 #include "content/renderer/media/webrtc/media_stream_track_metrics.h" | 11 #include "content/renderer/media/webrtc/media_stream_track_metrics.h" |
12 #include "content/renderer/media/webrtc/mock_peer_connection_dependency_factory.
h" | 12 #include "content/renderer/media/webrtc/mock_peer_connection_dependency_factory.
h" |
13 #include "testing/gmock/include/gmock/gmock.h" | 13 #include "testing/gmock/include/gmock/gmock.h" |
14 #include "testing/gtest/include/gtest/gtest.h" | 14 #include "testing/gtest/include/gtest/gtest.h" |
15 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" | 15 #include "third_party/webrtc/api/mediastreaminterface.h" |
16 | 16 |
17 using webrtc::AudioSourceInterface; | 17 using webrtc::AudioSourceInterface; |
18 using webrtc::AudioTrackInterface; | 18 using webrtc::AudioTrackInterface; |
19 using webrtc::AudioTrackSinkInterface; | 19 using webrtc::AudioTrackSinkInterface; |
20 using webrtc::MediaStreamInterface; | 20 using webrtc::MediaStreamInterface; |
21 using webrtc::ObserverInterface; | 21 using webrtc::ObserverInterface; |
22 using webrtc::PeerConnectionInterface; | 22 using webrtc::PeerConnectionInterface; |
23 using webrtc::VideoRendererInterface; | 23 using webrtc::VideoRendererInterface; |
24 using webrtc::VideoSourceInterface; | 24 using webrtc::VideoSourceInterface; |
25 using webrtc::VideoTrackInterface; | 25 using webrtc::VideoTrackInterface; |
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583 MediaStreamTrackMetrics::SENT_STREAM)); | 583 MediaStreamTrackMetrics::SENT_STREAM)); |
584 EXPECT_CALL(*metrics_, | 584 EXPECT_CALL(*metrics_, |
585 SendLifetimeMessage("video3", | 585 SendLifetimeMessage("video3", |
586 MediaStreamTrackMetrics::VIDEO_TRACK, | 586 MediaStreamTrackMetrics::VIDEO_TRACK, |
587 MediaStreamTrackMetrics::DISCONNECTED, | 587 MediaStreamTrackMetrics::DISCONNECTED, |
588 MediaStreamTrackMetrics::SENT_STREAM)); | 588 MediaStreamTrackMetrics::SENT_STREAM)); |
589 metrics_->RemoveStream(MediaStreamTrackMetrics::SENT_STREAM, stream_.get()); | 589 metrics_->RemoveStream(MediaStreamTrackMetrics::SENT_STREAM, stream_.get()); |
590 } | 590 } |
591 | 591 |
592 } // namespace content | 592 } // namespace content |
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