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Side by Side Diff: webrtc/api/peerconnection.cc

Issue 1748403002: Move RtcEventLog object from inside VoiceEngine to Call. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Undid unneccessary changes to rtp_rtcp module. Created 4 years, 9 months ago
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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
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25 #include "webrtc/api/mediastreamproxy.h" 25 #include "webrtc/api/mediastreamproxy.h"
26 #include "webrtc/api/mediastreamtrackproxy.h" 26 #include "webrtc/api/mediastreamtrackproxy.h"
27 #include "webrtc/api/remoteaudiosource.h" 27 #include "webrtc/api/remoteaudiosource.h"
28 #include "webrtc/api/remotevideocapturer.h" 28 #include "webrtc/api/remotevideocapturer.h"
29 #include "webrtc/api/rtpreceiver.h" 29 #include "webrtc/api/rtpreceiver.h"
30 #include "webrtc/api/rtpsender.h" 30 #include "webrtc/api/rtpsender.h"
31 #include "webrtc/api/streamcollection.h" 31 #include "webrtc/api/streamcollection.h"
32 #include "webrtc/api/videosource.h" 32 #include "webrtc/api/videosource.h"
33 #include "webrtc/api/videotrack.h" 33 #include "webrtc/api/videotrack.h"
34 #include "webrtc/base/arraysize.h" 34 #include "webrtc/base/arraysize.h"
35 #include "webrtc/base/bind.h"
35 #include "webrtc/base/logging.h" 36 #include "webrtc/base/logging.h"
36 #include "webrtc/base/stringencode.h" 37 #include "webrtc/base/stringencode.h"
37 #include "webrtc/base/stringutils.h" 38 #include "webrtc/base/stringutils.h"
38 #include "webrtc/base/trace_event.h" 39 #include "webrtc/base/trace_event.h"
40 #include "webrtc/call.h"
41 #include "webrtc/call/rtc_event_log.h"
39 #include "webrtc/media/sctp/sctpdataengine.h" 42 #include "webrtc/media/sctp/sctpdataengine.h"
40 #include "webrtc/p2p/client/basicportallocator.h" 43 #include "webrtc/p2p/client/basicportallocator.h"
41 #include "webrtc/pc/channelmanager.h" 44 #include "webrtc/pc/channelmanager.h"
42 #include "webrtc/system_wrappers/include/field_trial.h" 45 #include "webrtc/system_wrappers/include/field_trial.h"
43 46
44 namespace { 47 namespace {
45 48
46 using webrtc::DataChannel; 49 using webrtc::DataChannel;
47 using webrtc::MediaConstraintsInterface; 50 using webrtc::MediaConstraintsInterface;
48 using webrtc::MediaStreamInterface; 51 using webrtc::MediaStreamInterface;
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1219 kEnumCounterAddressFamily, kPeerConnection_IPv6, 1222 kEnumCounterAddressFamily, kPeerConnection_IPv6,
1220 kPeerConnectionAddressFamilyCounter_Max); 1223 kPeerConnectionAddressFamilyCounter_Max);
1221 } else { 1224 } else {
1222 uma_observer_->IncrementEnumCounter( 1225 uma_observer_->IncrementEnumCounter(
1223 kEnumCounterAddressFamily, kPeerConnection_IPv4, 1226 kEnumCounterAddressFamily, kPeerConnection_IPv4,
1224 kPeerConnectionAddressFamilyCounter_Max); 1227 kPeerConnectionAddressFamilyCounter_Max);
1225 } 1228 }
1226 } 1229 }
1227 } 1230 }
1228 1231
1232 bool PeerConnection::StartRtcEventLog(rtc::PlatformFile file,
1233 int64_t max_size_bytes) {
1234 return factory_->worker_thread()->Invoke<bool>(rtc::Bind(
1235 &PeerConnection::StartRtcEventLog_w, this, file, max_size_bytes));
1236 }
1237
1238 void PeerConnection::StopRtcEventLog() {
1239 factory_->worker_thread()->Invoke<void>(
1240 rtc::Bind(&PeerConnection::StopRtcEventLog_w, this));
1241 }
1242
1229 const SessionDescriptionInterface* PeerConnection::local_description() const { 1243 const SessionDescriptionInterface* PeerConnection::local_description() const {
1230 return session_->local_description(); 1244 return session_->local_description();
1231 } 1245 }
1232 1246
1233 const SessionDescriptionInterface* PeerConnection::remote_description() const { 1247 const SessionDescriptionInterface* PeerConnection::remote_description() const {
1234 return session_->remote_description(); 1248 return session_->remote_description();
1235 } 1249 }
1236 1250
1237 void PeerConnection::Close() { 1251 void PeerConnection::Close() {
1238 TRACE_EVENT0("webrtc", "PeerConnection::Close"); 1252 TRACE_EVENT0("webrtc", "PeerConnection::Close");
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2092 2106
2093 DataChannel* PeerConnection::FindDataChannelBySid(int sid) const { 2107 DataChannel* PeerConnection::FindDataChannelBySid(int sid) const {
2094 for (const auto& channel : sctp_data_channels_) { 2108 for (const auto& channel : sctp_data_channels_) {
2095 if (channel->id() == sid) { 2109 if (channel->id() == sid) {
2096 return channel; 2110 return channel;
2097 } 2111 }
2098 } 2112 }
2099 return nullptr; 2113 return nullptr;
2100 } 2114 }
2101 2115
2116 bool PeerConnection::StartRtcEventLog_w(rtc::PlatformFile file,
2117 int64_t max_size_bytes) {
2118 RTC_DCHECK(factory_->worker_thread()->IsCurrent());
2119 return media_controller_->call_w()->RtcEventLog()->StartLogging(
the sun 2016/03/21 13:03:08 Is Call::RtcEventLog used anywhere but here? If n
terelius 2016/03/21 20:47:46 +1
ivoc 2016/03/22 13:44:54 Good point, I changed it to a start/stop pair.
2120 file, max_size_bytes);
2121 }
2122
2123 void PeerConnection::StopRtcEventLog_w() {
2124 RTC_DCHECK(factory_->worker_thread()->IsCurrent());
2125 media_controller_->call_w()->RtcEventLog()->StopLogging();
2126 }
2102 } // namespace webrtc 2127 } // namespace webrtc
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