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1 /* | 1 /* |
2 * Copyright 2014 The WebRTC project authors. All Rights Reserved. | 2 * Copyright 2014 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 <stdio.h> | 11 #include <stdio.h> |
12 | 12 |
13 #include <algorithm> | 13 #include <algorithm> |
14 | 14 |
15 #include "webrtc/api/statscollector.h" | 15 #include "webrtc/api/statscollector.h" |
16 | 16 |
17 #include "testing/gmock/include/gmock/gmock.h" | 17 #include "testing/gmock/include/gmock/gmock.h" |
18 #include "testing/gtest/include/gtest/gtest.h" | 18 #include "testing/gtest/include/gtest/gtest.h" |
19 #include "webrtc/api/mediastream.h" | 19 #include "webrtc/api/mediastream.h" |
20 #include "webrtc/api/mediastreaminterface.h" | 20 #include "webrtc/api/mediastreaminterface.h" |
21 #include "webrtc/api/mediastreamtrack.h" | 21 #include "webrtc/api/mediastreamtrack.h" |
22 #include "webrtc/api/peerconnection.h" | 22 #include "webrtc/api/peerconnection.h" |
23 #include "webrtc/api/peerconnectionfactory.h" | 23 #include "webrtc/api/peerconnectionfactory.h" |
24 #include "webrtc/api/test/fakedatachannelprovider.h" | 24 #include "webrtc/api/test/fakedatachannelprovider.h" |
| 25 #include "webrtc/api/test/fakevideotracksource.h" |
25 #include "webrtc/api/videotrack.h" | 26 #include "webrtc/api/videotrack.h" |
26 #include "webrtc/base/base64.h" | 27 #include "webrtc/base/base64.h" |
27 #include "webrtc/base/fakesslidentity.h" | 28 #include "webrtc/base/fakesslidentity.h" |
28 #include "webrtc/base/gunit.h" | 29 #include "webrtc/base/gunit.h" |
29 #include "webrtc/base/network.h" | 30 #include "webrtc/base/network.h" |
30 #include "webrtc/media/base/fakemediaengine.h" | 31 #include "webrtc/media/base/fakemediaengine.h" |
31 #include "webrtc/p2p/base/faketransportcontroller.h" | 32 #include "webrtc/p2p/base/faketransportcontroller.h" |
32 #include "webrtc/pc/channelmanager.h" | 33 #include "webrtc/pc/channelmanager.h" |
33 | 34 |
34 using rtc::scoped_ptr; | 35 using rtc::scoped_ptr; |
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516 transport_stats.transport_name = kTransportName; | 517 transport_stats.transport_name = kTransportName; |
517 transport_stats.channel_stats.push_back(channel_stats); | 518 transport_stats.channel_stats.push_back(channel_stats); |
518 | 519 |
519 session_stats_.transport_stats[kTransportName] = transport_stats; | 520 session_stats_.transport_stats[kTransportName] = transport_stats; |
520 session_stats_.proxy_to_transport[vc_name] = kTransportName; | 521 session_stats_.proxy_to_transport[vc_name] = kTransportName; |
521 } | 522 } |
522 | 523 |
523 // Adds a outgoing video track with a given SSRC into the stats. | 524 // Adds a outgoing video track with a given SSRC into the stats. |
524 void AddOutgoingVideoTrackStats() { | 525 void AddOutgoingVideoTrackStats() { |
525 stream_ = webrtc::MediaStream::Create("streamlabel"); | 526 stream_ = webrtc::MediaStream::Create("streamlabel"); |
526 track_= webrtc::VideoTrack::Create(kLocalTrackId, NULL); | 527 track_ = webrtc::VideoTrack::Create(kLocalTrackId, |
| 528 webrtc::FakeVideoTrackSource::Create()); |
527 stream_->AddTrack(track_); | 529 stream_->AddTrack(track_); |
528 EXPECT_CALL(session_, GetLocalTrackIdBySsrc(kSsrcOfTrack, _)) | 530 EXPECT_CALL(session_, GetLocalTrackIdBySsrc(kSsrcOfTrack, _)) |
529 .WillRepeatedly(DoAll(SetArgPointee<1>(kLocalTrackId), Return(true))); | 531 .WillRepeatedly(DoAll(SetArgPointee<1>(kLocalTrackId), Return(true))); |
530 } | 532 } |
531 | 533 |
532 // Adds a incoming video track with a given SSRC into the stats. | 534 // Adds a incoming video track with a given SSRC into the stats. |
533 void AddIncomingVideoTrackStats() { | 535 void AddIncomingVideoTrackStats() { |
534 stream_ = webrtc::MediaStream::Create("streamlabel"); | 536 stream_ = webrtc::MediaStream::Create("streamlabel"); |
535 track_= webrtc::VideoTrack::Create(kRemoteTrackId, NULL); | 537 track_ = webrtc::VideoTrack::Create(kRemoteTrackId, |
| 538 webrtc::FakeVideoTrackSource::Create()); |
536 stream_->AddTrack(track_); | 539 stream_->AddTrack(track_); |
537 EXPECT_CALL(session_, GetRemoteTrackIdBySsrc(kSsrcOfTrack, _)) | 540 EXPECT_CALL(session_, GetRemoteTrackIdBySsrc(kSsrcOfTrack, _)) |
538 .WillRepeatedly(DoAll(SetArgPointee<1>(kRemoteTrackId), Return(true))); | 541 .WillRepeatedly(DoAll(SetArgPointee<1>(kRemoteTrackId), Return(true))); |
539 } | 542 } |
540 | 543 |
541 // Adds a outgoing audio track with a given SSRC into the stats. | 544 // Adds a outgoing audio track with a given SSRC into the stats. |
542 void AddOutgoingAudioTrackStats() { | 545 void AddOutgoingAudioTrackStats() { |
543 if (stream_ == NULL) | 546 if (stream_ == NULL) |
544 stream_ = webrtc::MediaStream::Create("streamlabel"); | 547 stream_ = webrtc::MediaStream::Create("streamlabel"); |
545 | 548 |
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1731 cricket::VoiceSenderInfo new_voice_sender_info; | 1734 cricket::VoiceSenderInfo new_voice_sender_info; |
1732 InitVoiceSenderInfo(&new_voice_sender_info); | 1735 InitVoiceSenderInfo(&new_voice_sender_info); |
1733 cricket::VoiceMediaInfo new_stats_read; | 1736 cricket::VoiceMediaInfo new_stats_read; |
1734 reports.clear(); | 1737 reports.clear(); |
1735 SetupAndVerifyAudioTrackStats( | 1738 SetupAndVerifyAudioTrackStats( |
1736 new_audio_track.get(), stream_.get(), &stats, &voice_channel, kVcName, | 1739 new_audio_track.get(), stream_.get(), &stats, &voice_channel, kVcName, |
1737 media_channel, &new_voice_sender_info, NULL, &new_stats_read, &reports); | 1740 media_channel, &new_voice_sender_info, NULL, &new_stats_read, &reports); |
1738 } | 1741 } |
1739 | 1742 |
1740 } // namespace webrtc | 1743 } // namespace webrtc |
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