| OLD | NEW |
| 1 /* | 1 /* |
| 2 * Copyright 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright 2012 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 "webrtc/api/peerconnection.h" | 11 #include "webrtc/api/peerconnection.h" |
| 12 | 12 |
| 13 #include <algorithm> | 13 #include <algorithm> |
| 14 #include <cctype> // for isdigit | 14 #include <cctype> // for isdigit |
| 15 #include <utility> | 15 #include <utility> |
| 16 #include <vector> | 16 #include <vector> |
| 17 | 17 |
| 18 #include "talk/session/media/channelmanager.h" | |
| 19 #include "webrtc/api/audiotrack.h" | 18 #include "webrtc/api/audiotrack.h" |
| 20 #include "webrtc/api/dtmfsender.h" | 19 #include "webrtc/api/dtmfsender.h" |
| 21 #include "webrtc/api/jsepicecandidate.h" | 20 #include "webrtc/api/jsepicecandidate.h" |
| 22 #include "webrtc/api/jsepsessiondescription.h" | 21 #include "webrtc/api/jsepsessiondescription.h" |
| 23 #include "webrtc/api/mediaconstraintsinterface.h" | 22 #include "webrtc/api/mediaconstraintsinterface.h" |
| 24 #include "webrtc/api/mediastream.h" | 23 #include "webrtc/api/mediastream.h" |
| 25 #include "webrtc/api/mediastreamobserver.h" | 24 #include "webrtc/api/mediastreamobserver.h" |
| 26 #include "webrtc/api/mediastreamproxy.h" | 25 #include "webrtc/api/mediastreamproxy.h" |
| 27 #include "webrtc/api/mediastreamtrackproxy.h" | 26 #include "webrtc/api/mediastreamtrackproxy.h" |
| 28 #include "webrtc/api/remoteaudiosource.h" | 27 #include "webrtc/api/remoteaudiosource.h" |
| 29 #include "webrtc/api/remotevideocapturer.h" | 28 #include "webrtc/api/remotevideocapturer.h" |
| 30 #include "webrtc/api/rtpreceiver.h" | 29 #include "webrtc/api/rtpreceiver.h" |
| 31 #include "webrtc/api/rtpsender.h" | 30 #include "webrtc/api/rtpsender.h" |
| 32 #include "webrtc/api/streamcollection.h" | 31 #include "webrtc/api/streamcollection.h" |
| 33 #include "webrtc/api/videosource.h" | 32 #include "webrtc/api/videosource.h" |
| 34 #include "webrtc/api/videotrack.h" | 33 #include "webrtc/api/videotrack.h" |
| 35 #include "webrtc/base/arraysize.h" | 34 #include "webrtc/base/arraysize.h" |
| 36 #include "webrtc/base/logging.h" | 35 #include "webrtc/base/logging.h" |
| 37 #include "webrtc/base/stringencode.h" | 36 #include "webrtc/base/stringencode.h" |
| 38 #include "webrtc/base/stringutils.h" | 37 #include "webrtc/base/stringutils.h" |
| 39 #include "webrtc/base/trace_event.h" | 38 #include "webrtc/base/trace_event.h" |
| 40 #include "webrtc/media/sctp/sctpdataengine.h" | 39 #include "webrtc/media/sctp/sctpdataengine.h" |
| 41 #include "webrtc/p2p/client/basicportallocator.h" | 40 #include "webrtc/p2p/client/basicportallocator.h" |
| 41 #include "webrtc/pc/channelmanager.h" |
| 42 #include "webrtc/system_wrappers/include/field_trial.h" | 42 #include "webrtc/system_wrappers/include/field_trial.h" |
| 43 | 43 |
| 44 namespace { | 44 namespace { |
| 45 | 45 |
| 46 using webrtc::DataChannel; | 46 using webrtc::DataChannel; |
| 47 using webrtc::MediaConstraintsInterface; | 47 using webrtc::MediaConstraintsInterface; |
| 48 using webrtc::MediaStreamInterface; | 48 using webrtc::MediaStreamInterface; |
| 49 using webrtc::PeerConnectionInterface; | 49 using webrtc::PeerConnectionInterface; |
| 50 using webrtc::RtpSenderInterface; | 50 using webrtc::RtpSenderInterface; |
| 51 using webrtc::StreamCollection; | 51 using webrtc::StreamCollection; |
| (...skipping 2013 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
| 2065 DataChannel* PeerConnection::FindDataChannelBySid(int sid) const { | 2065 DataChannel* PeerConnection::FindDataChannelBySid(int sid) const { |
| 2066 for (const auto& channel : sctp_data_channels_) { | 2066 for (const auto& channel : sctp_data_channels_) { |
| 2067 if (channel->id() == sid) { | 2067 if (channel->id() == sid) { |
| 2068 return channel; | 2068 return channel; |
| 2069 } | 2069 } |
| 2070 } | 2070 } |
| 2071 return nullptr; | 2071 return nullptr; |
| 2072 } | 2072 } |
| 2073 | 2073 |
| 2074 } // namespace webrtc | 2074 } // namespace webrtc |
| OLD | NEW |