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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, |
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44 #include "talk/app/webrtc/rtpsender.h" | 44 #include "talk/app/webrtc/rtpsender.h" |
45 #include "talk/app/webrtc/streamcollection.h" | 45 #include "talk/app/webrtc/streamcollection.h" |
46 #include "talk/app/webrtc/videosource.h" | 46 #include "talk/app/webrtc/videosource.h" |
47 #include "talk/app/webrtc/videotrack.h" | 47 #include "talk/app/webrtc/videotrack.h" |
48 #include "talk/media/sctp/sctpdataengine.h" | 48 #include "talk/media/sctp/sctpdataengine.h" |
49 #include "webrtc/p2p/client/basicportallocator.h" | 49 #include "webrtc/p2p/client/basicportallocator.h" |
50 #include "talk/session/media/channelmanager.h" | 50 #include "talk/session/media/channelmanager.h" |
51 #include "webrtc/base/logging.h" | 51 #include "webrtc/base/logging.h" |
52 #include "webrtc/base/stringencode.h" | 52 #include "webrtc/base/stringencode.h" |
53 #include "webrtc/base/stringutils.h" | 53 #include "webrtc/base/stringutils.h" |
54 #include "webrtc/system_wrappers/interface/field_trial.h" | 54 #include "webrtc/system_wrappers/include/field_trial.h" |
55 | 55 |
56 namespace { | 56 namespace { |
57 | 57 |
58 using webrtc::DataChannel; | 58 using webrtc::DataChannel; |
59 using webrtc::MediaConstraintsInterface; | 59 using webrtc::MediaConstraintsInterface; |
60 using webrtc::MediaStreamInterface; | 60 using webrtc::MediaStreamInterface; |
61 using webrtc::PeerConnectionInterface; | 61 using webrtc::PeerConnectionInterface; |
62 using webrtc::StreamCollection; | 62 using webrtc::StreamCollection; |
63 using webrtc::StunConfigurations; | 63 using webrtc::StunConfigurations; |
64 using webrtc::TurnConfigurations; | 64 using webrtc::TurnConfigurations; |
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1966 DataChannel* PeerConnection::FindDataChannelBySid(int sid) const { | 1966 DataChannel* PeerConnection::FindDataChannelBySid(int sid) const { |
1967 for (const auto& channel : sctp_data_channels_) { | 1967 for (const auto& channel : sctp_data_channels_) { |
1968 if (channel->id() == sid) { | 1968 if (channel->id() == sid) { |
1969 return channel; | 1969 return channel; |
1970 } | 1970 } |
1971 } | 1971 } |
1972 return nullptr; | 1972 return nullptr; |
1973 } | 1973 } |
1974 | 1974 |
1975 } // namespace webrtc | 1975 } // namespace webrtc |
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