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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" | 18 #include "webrtc/api/jsepicecandidate.h" |
21 #include "webrtc/api/jsepsessiondescription.h" | 19 #include "webrtc/api/jsepsessiondescription.h" |
22 #include "webrtc/api/mediaconstraintsinterface.h" | 20 #include "webrtc/api/mediaconstraintsinterface.h" |
23 #include "webrtc/api/mediastream.h" | |
24 #include "webrtc/api/mediastreamobserver.h" | |
25 #include "webrtc/api/mediastreamproxy.h" | 21 #include "webrtc/api/mediastreamproxy.h" |
26 #include "webrtc/api/mediastreamtrackproxy.h" | 22 #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" | 23 #include "webrtc/base/arraysize.h" |
34 #include "webrtc/base/bind.h" | 24 #include "webrtc/base/bind.h" |
35 #include "webrtc/base/checks.h" | 25 #include "webrtc/base/checks.h" |
36 #include "webrtc/base/logging.h" | 26 #include "webrtc/base/logging.h" |
37 #include "webrtc/base/stringencode.h" | 27 #include "webrtc/base/stringencode.h" |
38 #include "webrtc/base/stringutils.h" | 28 #include "webrtc/base/stringutils.h" |
39 #include "webrtc/base/trace_event.h" | 29 #include "webrtc/base/trace_event.h" |
40 #include "webrtc/call/call.h" | 30 #include "webrtc/call/call.h" |
41 #include "webrtc/logging/rtc_event_log/rtc_event_log.h" | 31 #include "webrtc/logging/rtc_event_log/rtc_event_log.h" |
42 #include "webrtc/media/sctp/sctptransport.h" | 32 #include "webrtc/media/sctp/sctptransport.h" |
| 33 #include "webrtc/pc/audiotrack.h" |
43 #include "webrtc/pc/channelmanager.h" | 34 #include "webrtc/pc/channelmanager.h" |
| 35 #include "webrtc/pc/dtmfsender.h" |
| 36 #include "webrtc/pc/mediastream.h" |
| 37 #include "webrtc/pc/mediastreamobserver.h" |
| 38 #include "webrtc/pc/remoteaudiosource.h" |
| 39 #include "webrtc/pc/rtpreceiver.h" |
| 40 #include "webrtc/pc/rtpsender.h" |
| 41 #include "webrtc/pc/streamcollection.h" |
| 42 #include "webrtc/pc/videocapturertracksource.h" |
| 43 #include "webrtc/pc/videotrack.h" |
| 44 #include "webrtc/system_wrappers/include/clock.h" |
44 #include "webrtc/system_wrappers/include/field_trial.h" | 45 #include "webrtc/system_wrappers/include/field_trial.h" |
45 | 46 |
46 namespace { | 47 namespace { |
47 | 48 |
48 using webrtc::DataChannel; | 49 using webrtc::DataChannel; |
49 using webrtc::MediaConstraintsInterface; | 50 using webrtc::MediaConstraintsInterface; |
50 using webrtc::MediaStreamInterface; | 51 using webrtc::MediaStreamInterface; |
51 using webrtc::PeerConnectionInterface; | 52 using webrtc::PeerConnectionInterface; |
52 using webrtc::RTCError; | 53 using webrtc::RTCError; |
53 using webrtc::RTCErrorType; | 54 using webrtc::RTCErrorType; |
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2564 | 2565 |
2565 bool PeerConnection::StartRtcEventLog_w(rtc::PlatformFile file, | 2566 bool PeerConnection::StartRtcEventLog_w(rtc::PlatformFile file, |
2566 int64_t max_size_bytes) { | 2567 int64_t max_size_bytes) { |
2567 return event_log_->StartLogging(file, max_size_bytes); | 2568 return event_log_->StartLogging(file, max_size_bytes); |
2568 } | 2569 } |
2569 | 2570 |
2570 void PeerConnection::StopRtcEventLog_w() { | 2571 void PeerConnection::StopRtcEventLog_w() { |
2571 event_log_->StopLogging(); | 2572 event_log_->StopLogging(); |
2572 } | 2573 } |
2573 } // namespace webrtc | 2574 } // namespace webrtc |
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