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| 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/pc/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 "webrtc/api/audiotrack.h" | |
| 19 #include "webrtc/api/dtmfsender.h" | |
| 20 #include "webrtc/api/jsepicecandidate.h" | |
| 21 #include "webrtc/api/jsepsessiondescription.h" | |
| 22 #include "webrtc/api/mediaconstraintsinterface.h" | 18 #include "webrtc/api/mediaconstraintsinterface.h" |
| 23 #include "webrtc/api/mediastream.h" | |
| 24 #include "webrtc/api/mediastreamobserver.h" | |
| 25 #include "webrtc/api/mediastreamproxy.h" | 19 #include "webrtc/api/mediastreamproxy.h" |
| 26 #include "webrtc/api/mediastreamtrackproxy.h" | 20 #include "webrtc/api/mediastreamtrackproxy.h" |
| 27 #include "webrtc/api/remoteaudiosource.h" | |
| 28 #include "webrtc/api/rtpreceiver.h" | |
| 29 #include "webrtc/api/rtpsender.h" | |
| 30 #include "webrtc/api/streamcollection.h" | |
| 31 #include "webrtc/api/videocapturertracksource.h" | |
| 32 #include "webrtc/api/videotrack.h" | |
| 33 #include "webrtc/base/arraysize.h" | 21 #include "webrtc/base/arraysize.h" |
| 34 #include "webrtc/base/bind.h" | 22 #include "webrtc/base/bind.h" |
| 35 #include "webrtc/base/logging.h" | 23 #include "webrtc/base/logging.h" |
| 36 #include "webrtc/base/stringencode.h" | 24 #include "webrtc/base/stringencode.h" |
| 37 #include "webrtc/base/stringutils.h" | 25 #include "webrtc/base/stringutils.h" |
| 38 #include "webrtc/base/trace_event.h" | 26 #include "webrtc/base/trace_event.h" |
| 39 #include "webrtc/call/call.h" | 27 #include "webrtc/call/call.h" |
| 40 #include "webrtc/logging/rtc_event_log/rtc_event_log.h" | 28 #include "webrtc/logging/rtc_event_log/rtc_event_log.h" |
| 41 #include "webrtc/media/sctp/sctpdataengine.h" | 29 #include "webrtc/media/sctp/sctpdataengine.h" |
| 30 #include "webrtc/pc/audiotrack.h" |
| 42 #include "webrtc/pc/channelmanager.h" | 31 #include "webrtc/pc/channelmanager.h" |
| 32 #include "webrtc/pc/dtmfsender.h" |
| 33 #include "webrtc/pc/jsepicecandidate.h" |
| 34 #include "webrtc/pc/jsepsessiondescription.h" |
| 35 #include "webrtc/pc/mediastream.h" |
| 36 #include "webrtc/pc/mediastreamobserver.h" |
| 37 #include "webrtc/pc/remoteaudiosource.h" |
| 38 #include "webrtc/pc/rtpreceiver.h" |
| 39 #include "webrtc/pc/rtpsender.h" |
| 40 #include "webrtc/pc/streamcollection.h" |
| 41 #include "webrtc/pc/videocapturertracksource.h" |
| 42 #include "webrtc/pc/videotrack.h" |
| 43 #include "webrtc/system_wrappers/include/clock.h" | 43 #include "webrtc/system_wrappers/include/clock.h" |
| 44 #include "webrtc/system_wrappers/include/field_trial.h" | 44 #include "webrtc/system_wrappers/include/field_trial.h" |
| 45 | 45 |
| 46 namespace { | 46 namespace { |
| 47 | 47 |
| 48 using webrtc::DataChannel; | 48 using webrtc::DataChannel; |
| 49 using webrtc::MediaConstraintsInterface; | 49 using webrtc::MediaConstraintsInterface; |
| 50 using webrtc::MediaStreamInterface; | 50 using webrtc::MediaStreamInterface; |
| 51 using webrtc::PeerConnectionInterface; | 51 using webrtc::PeerConnectionInterface; |
| 52 using webrtc::RtpSenderInternal; | 52 using webrtc::RtpSenderInternal; |
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| 2400 | 2400 |
| 2401 bool PeerConnection::StartRtcEventLog_w(rtc::PlatformFile file, | 2401 bool PeerConnection::StartRtcEventLog_w(rtc::PlatformFile file, |
| 2402 int64_t max_size_bytes) { | 2402 int64_t max_size_bytes) { |
| 2403 return event_log_->StartLogging(file, max_size_bytes); | 2403 return event_log_->StartLogging(file, max_size_bytes); |
| 2404 } | 2404 } |
| 2405 | 2405 |
| 2406 void PeerConnection::StopRtcEventLog_w() { | 2406 void PeerConnection::StopRtcEventLog_w() { |
| 2407 event_log_->StopLogging(); | 2407 event_log_->StopLogging(); |
| 2408 } | 2408 } |
| 2409 } // namespace webrtc | 2409 } // namespace webrtc |
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