OLD | NEW |
1 // Copyright (c) 2012 The Chromium Authors. All rights reserved. | 1 // Copyright (c) 2012 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 "content/renderer/media/rtc_peer_connection_handler.h" | 5 #include "content/renderer/media/rtc_peer_connection_handler.h" |
6 | 6 |
7 #include <stddef.h> | 7 #include <stddef.h> |
8 | 8 |
9 #include <memory> | 9 #include <memory> |
10 #include <set> | 10 #include <set> |
(...skipping 14 matching lines...) Expand all Loading... |
25 #include "content/renderer/media/media_stream_video_track.h" | 25 #include "content/renderer/media/media_stream_video_track.h" |
26 #include "content/renderer/media/mock_audio_device_factory.h" | 26 #include "content/renderer/media/mock_audio_device_factory.h" |
27 #include "content/renderer/media/mock_constraint_factory.h" | 27 #include "content/renderer/media/mock_constraint_factory.h" |
28 #include "content/renderer/media/mock_data_channel_impl.h" | 28 #include "content/renderer/media/mock_data_channel_impl.h" |
29 #include "content/renderer/media/mock_media_stream_video_source.h" | 29 #include "content/renderer/media/mock_media_stream_video_source.h" |
30 #include "content/renderer/media/mock_peer_connection_impl.h" | 30 #include "content/renderer/media/mock_peer_connection_impl.h" |
31 #include "content/renderer/media/mock_web_rtc_peer_connection_handler_client.h" | 31 #include "content/renderer/media/mock_web_rtc_peer_connection_handler_client.h" |
32 #include "content/renderer/media/peer_connection_tracker.h" | 32 #include "content/renderer/media/peer_connection_tracker.h" |
33 #include "content/renderer/media/webrtc/mock_peer_connection_dependency_factory.
h" | 33 #include "content/renderer/media/webrtc/mock_peer_connection_dependency_factory.
h" |
34 #include "content/renderer/media/webrtc/processed_local_audio_source.h" | 34 #include "content/renderer/media/webrtc/processed_local_audio_source.h" |
| 35 #include "content/renderer/media/webrtc/rtc_stats.h" |
35 #include "testing/gmock/include/gmock/gmock.h" | 36 #include "testing/gmock/include/gmock/gmock.h" |
36 #include "testing/gtest/include/gtest/gtest.h" | 37 #include "testing/gtest/include/gtest/gtest.h" |
37 #include "third_party/WebKit/public/platform/WebMediaConstraints.h" | 38 #include "third_party/WebKit/public/platform/WebMediaConstraints.h" |
38 #include "third_party/WebKit/public/platform/WebMediaStream.h" | 39 #include "third_party/WebKit/public/platform/WebMediaStream.h" |
39 #include "third_party/WebKit/public/platform/WebMediaStreamSource.h" | 40 #include "third_party/WebKit/public/platform/WebMediaStreamSource.h" |
40 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h" | 41 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h" |
41 #include "third_party/WebKit/public/platform/WebRTCConfiguration.h" | 42 #include "third_party/WebKit/public/platform/WebRTCConfiguration.h" |
42 #include "third_party/WebKit/public/platform/WebRTCDTMFSenderHandler.h" | 43 #include "third_party/WebKit/public/platform/WebRTCDTMFSenderHandler.h" |
43 #include "third_party/WebKit/public/platform/WebRTCDataChannelHandler.h" | 44 #include "third_party/WebKit/public/platform/WebRTCDataChannelHandler.h" |
44 #include "third_party/WebKit/public/platform/WebRTCDataChannelInit.h" | 45 #include "third_party/WebKit/public/platform/WebRTCDataChannelInit.h" |
(...skipping 620 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
665 new rtc::RefCountedObject<MockRTCStatsRequest>()); | 666 new rtc::RefCountedObject<MockRTCStatsRequest>()); |
666 request->setSelector(component); | 667 request->setSelector(component); |
667 pc_handler_->getStats(request.get()); | 668 pc_handler_->getStats(request.get()); |
668 base::RunLoop().RunUntilIdle(); | 669 base::RunLoop().RunUntilIdle(); |
669 EXPECT_EQ(0, request->result()->report_count()); | 670 EXPECT_EQ(0, request->result()->report_count()); |
670 | 671 |
671 StopAllTracks(local_stream); | 672 StopAllTracks(local_stream); |
672 } | 673 } |
673 | 674 |
674 TEST_F(RTCPeerConnectionHandlerTest, GetRTCStats) { | 675 TEST_F(RTCPeerConnectionHandlerTest, GetRTCStats) { |
| 676 WhitelistStatsForTesting(webrtc::RTCTestStats::kType); |
| 677 |
675 rtc::scoped_refptr<webrtc::RTCStatsReport> report = | 678 rtc::scoped_refptr<webrtc::RTCStatsReport> report = |
676 webrtc::RTCStatsReport::Create(); | 679 webrtc::RTCStatsReport::Create(); |
677 | 680 |
678 report->AddStats(std::unique_ptr<const webrtc::RTCStats>( | 681 report->AddStats(std::unique_ptr<const webrtc::RTCStats>( |
679 new webrtc::RTCTestStats("RTCUndefinedStats", 1000))); | 682 new webrtc::RTCTestStats("RTCUndefinedStats", 1000))); |
680 | 683 |
681 std::unique_ptr<webrtc::RTCTestStats> stats_defined_members( | 684 std::unique_ptr<webrtc::RTCTestStats> stats_defined_members( |
682 new webrtc::RTCTestStats("RTCDefinedStats", 2000)); | 685 new webrtc::RTCTestStats("RTCDefinedStats", 2000)); |
683 stats_defined_members->m_bool = true; | 686 stats_defined_members->m_bool = true; |
684 stats_defined_members->m_int32 = 42; | 687 stats_defined_members->m_int32 = 42; |
(...skipping 528 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
1213 testing::Ref(tracks[0]))); | 1216 testing::Ref(tracks[0]))); |
1214 | 1217 |
1215 std::unique_ptr<blink::WebRTCDTMFSenderHandler> sender( | 1218 std::unique_ptr<blink::WebRTCDTMFSenderHandler> sender( |
1216 pc_handler_->createDTMFSender(tracks[0])); | 1219 pc_handler_->createDTMFSender(tracks[0])); |
1217 EXPECT_TRUE(sender.get()); | 1220 EXPECT_TRUE(sender.get()); |
1218 | 1221 |
1219 StopAllTracks(local_stream); | 1222 StopAllTracks(local_stream); |
1220 } | 1223 } |
1221 | 1224 |
1222 } // namespace content | 1225 } // namespace content |
OLD | NEW |