Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(114)

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

Issue 2515653002: Convert rtc_event_log from webrtc::Clock to rtc::TimeMicros. (Closed)
Patch Set: Rebase. Created 3 years, 12 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
« no previous file with comments | « no previous file | webrtc/base/fakeclock.h » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
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
(...skipping 22 matching lines...) Expand all
33 #include "webrtc/base/arraysize.h" 33 #include "webrtc/base/arraysize.h"
34 #include "webrtc/base/bind.h" 34 #include "webrtc/base/bind.h"
35 #include "webrtc/base/logging.h" 35 #include "webrtc/base/logging.h"
36 #include "webrtc/base/stringencode.h" 36 #include "webrtc/base/stringencode.h"
37 #include "webrtc/base/stringutils.h" 37 #include "webrtc/base/stringutils.h"
38 #include "webrtc/base/trace_event.h" 38 #include "webrtc/base/trace_event.h"
39 #include "webrtc/call/call.h" 39 #include "webrtc/call/call.h"
40 #include "webrtc/logging/rtc_event_log/rtc_event_log.h" 40 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
41 #include "webrtc/media/sctp/sctpdataengine.h" 41 #include "webrtc/media/sctp/sctpdataengine.h"
42 #include "webrtc/pc/channelmanager.h" 42 #include "webrtc/pc/channelmanager.h"
43 #include "webrtc/system_wrappers/include/clock.h"
44 #include "webrtc/system_wrappers/include/field_trial.h" 43 #include "webrtc/system_wrappers/include/field_trial.h"
45 44
46 namespace { 45 namespace {
47 46
48 using webrtc::DataChannel; 47 using webrtc::DataChannel;
49 using webrtc::MediaConstraintsInterface; 48 using webrtc::MediaConstraintsInterface;
50 using webrtc::MediaStreamInterface; 49 using webrtc::MediaStreamInterface;
51 using webrtc::PeerConnectionInterface; 50 using webrtc::PeerConnectionInterface;
52 using webrtc::RtpSenderInternal; 51 using webrtc::RtpSenderInternal;
53 using webrtc::RtpSenderInterface; 52 using webrtc::RtpSenderInterface;
(...skipping 529 matching lines...) Expand 10 before | Expand all | Expand 10 after
583 return true; 582 return true;
584 } 583 }
585 584
586 PeerConnection::PeerConnection(PeerConnectionFactory* factory) 585 PeerConnection::PeerConnection(PeerConnectionFactory* factory)
587 : factory_(factory), 586 : factory_(factory),
588 observer_(NULL), 587 observer_(NULL),
589 uma_observer_(NULL), 588 uma_observer_(NULL),
590 signaling_state_(kStable), 589 signaling_state_(kStable),
591 ice_connection_state_(kIceConnectionNew), 590 ice_connection_state_(kIceConnectionNew),
592 ice_gathering_state_(kIceGatheringNew), 591 ice_gathering_state_(kIceGatheringNew),
593 event_log_(RtcEventLog::Create(webrtc::Clock::GetRealTimeClock())), 592 event_log_(RtcEventLog::Create()),
594 rtcp_cname_(GenerateRtcpCname()), 593 rtcp_cname_(GenerateRtcpCname()),
595 local_streams_(StreamCollection::Create()), 594 local_streams_(StreamCollection::Create()),
596 remote_streams_(StreamCollection::Create()) {} 595 remote_streams_(StreamCollection::Create()) {}
597 596
598 PeerConnection::~PeerConnection() { 597 PeerConnection::~PeerConnection() {
599 TRACE_EVENT0("webrtc", "PeerConnection::~PeerConnection"); 598 TRACE_EVENT0("webrtc", "PeerConnection::~PeerConnection");
600 RTC_DCHECK(signaling_thread()->IsCurrent()); 599 RTC_DCHECK(signaling_thread()->IsCurrent());
601 // Need to detach RTP senders/receivers from WebRtcSession, 600 // Need to detach RTP senders/receivers from WebRtcSession,
602 // since it's about to be destroyed. 601 // since it's about to be destroyed.
603 for (const auto& sender : senders_) { 602 for (const auto& sender : senders_) {
(...skipping 1800 matching lines...) Expand 10 before | Expand all | Expand 10 after
2404 2403
2405 bool PeerConnection::StartRtcEventLog_w(rtc::PlatformFile file, 2404 bool PeerConnection::StartRtcEventLog_w(rtc::PlatformFile file,
2406 int64_t max_size_bytes) { 2405 int64_t max_size_bytes) {
2407 return event_log_->StartLogging(file, max_size_bytes); 2406 return event_log_->StartLogging(file, max_size_bytes);
2408 } 2407 }
2409 2408
2410 void PeerConnection::StopRtcEventLog_w() { 2409 void PeerConnection::StopRtcEventLog_w() {
2411 event_log_->StopLogging(); 2410 event_log_->StopLogging();
2412 } 2411 }
2413 } // namespace webrtc 2412 } // namespace webrtc
OLDNEW
« no previous file with comments | « no previous file | webrtc/base/fakeclock.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698