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Issue 1505253004: Support for remote audio into tracks (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Address comments Created 5 years ago
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1 /* 1 /*
2 * libjingle 2 * libjingle
3 * Copyright 2014 Google Inc. 3 * Copyright 2014 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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35 #include "talk/app/webrtc/peerconnectionfactory.h" 35 #include "talk/app/webrtc/peerconnectionfactory.h"
36 #include "talk/app/webrtc/mediastream.h" 36 #include "talk/app/webrtc/mediastream.h"
37 #include "talk/app/webrtc/mediastreaminterface.h" 37 #include "talk/app/webrtc/mediastreaminterface.h"
38 #include "talk/app/webrtc/mediastreamtrack.h" 38 #include "talk/app/webrtc/mediastreamtrack.h"
39 #include "talk/app/webrtc/test/fakedatachannelprovider.h" 39 #include "talk/app/webrtc/test/fakedatachannelprovider.h"
40 #include "talk/app/webrtc/videotrack.h" 40 #include "talk/app/webrtc/videotrack.h"
41 #include "talk/media/base/fakemediaengine.h" 41 #include "talk/media/base/fakemediaengine.h"
42 #include "talk/session/media/channelmanager.h" 42 #include "talk/session/media/channelmanager.h"
43 #include "testing/gmock/include/gmock/gmock.h" 43 #include "testing/gmock/include/gmock/gmock.h"
44 #include "testing/gtest/include/gtest/gtest.h" 44 #include "testing/gtest/include/gtest/gtest.h"
45 #include "webrtc/audio/audio_sink.h"
45 #include "webrtc/base/base64.h" 46 #include "webrtc/base/base64.h"
46 #include "webrtc/base/fakesslidentity.h" 47 #include "webrtc/base/fakesslidentity.h"
47 #include "webrtc/base/gunit.h" 48 #include "webrtc/base/gunit.h"
48 #include "webrtc/base/network.h" 49 #include "webrtc/base/network.h"
49 #include "webrtc/p2p/base/faketransportcontroller.h" 50 #include "webrtc/p2p/base/faketransportcontroller.h"
50 51
51 using rtc::scoped_ptr; 52 using rtc::scoped_ptr;
52 using testing::_; 53 using testing::_;
53 using testing::DoAll; 54 using testing::DoAll;
54 using testing::Field; 55 using testing::Field;
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1740 cricket::VoiceSenderInfo new_voice_sender_info; 1741 cricket::VoiceSenderInfo new_voice_sender_info;
1741 InitVoiceSenderInfo(&new_voice_sender_info); 1742 InitVoiceSenderInfo(&new_voice_sender_info);
1742 cricket::VoiceMediaInfo new_stats_read; 1743 cricket::VoiceMediaInfo new_stats_read;
1743 reports.clear(); 1744 reports.clear();
1744 SetupAndVerifyAudioTrackStats( 1745 SetupAndVerifyAudioTrackStats(
1745 new_audio_track.get(), stream_.get(), &stats, &voice_channel, kVcName, 1746 new_audio_track.get(), stream_.get(), &stats, &voice_channel, kVcName,
1746 media_channel, &new_voice_sender_info, NULL, &new_stats_read, &reports); 1747 media_channel, &new_voice_sender_info, NULL, &new_stats_read, &reports);
1747 } 1748 }
1748 1749
1749 } // namespace webrtc 1750 } // namespace webrtc
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