Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(774)

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

Issue 2550273003: Moved call.h and most of api/call/* into call/ (Closed)
Patch Set: Added <set> include after presubmit complained. Created 4 years 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 | « webrtc/api/peerconnection.cc ('k') | webrtc/audio/BUILD.gn » ('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 12 matching lines...) Expand all
23 #include "webrtc/api/peerconnectioninterface.h" 23 #include "webrtc/api/peerconnectioninterface.h"
24 #include "webrtc/api/sctputils.h" 24 #include "webrtc/api/sctputils.h"
25 #include "webrtc/api/webrtcsessiondescriptionfactory.h" 25 #include "webrtc/api/webrtcsessiondescriptionfactory.h"
26 #include "webrtc/base/basictypes.h" 26 #include "webrtc/base/basictypes.h"
27 #include "webrtc/base/bind.h" 27 #include "webrtc/base/bind.h"
28 #include "webrtc/base/checks.h" 28 #include "webrtc/base/checks.h"
29 #include "webrtc/base/helpers.h" 29 #include "webrtc/base/helpers.h"
30 #include "webrtc/base/logging.h" 30 #include "webrtc/base/logging.h"
31 #include "webrtc/base/stringencode.h" 31 #include "webrtc/base/stringencode.h"
32 #include "webrtc/base/stringutils.h" 32 #include "webrtc/base/stringutils.h"
33 #include "webrtc/call.h" 33 #include "webrtc/call/call.h"
34 #include "webrtc/media/base/mediaconstants.h" 34 #include "webrtc/media/base/mediaconstants.h"
35 #include "webrtc/media/base/videocapturer.h" 35 #include "webrtc/media/base/videocapturer.h"
36 #include "webrtc/p2p/base/portallocator.h" 36 #include "webrtc/p2p/base/portallocator.h"
37 #include "webrtc/p2p/base/transportchannel.h" 37 #include "webrtc/p2p/base/transportchannel.h"
38 #include "webrtc/pc/channel.h" 38 #include "webrtc/pc/channel.h"
39 #include "webrtc/pc/channelmanager.h" 39 #include "webrtc/pc/channelmanager.h"
40 #include "webrtc/pc/mediasession.h" 40 #include "webrtc/pc/mediasession.h"
41 41
42 #ifdef HAVE_QUIC 42 #ifdef HAVE_QUIC
43 #include "webrtc/p2p/quic/quictransportchannel.h" 43 #include "webrtc/p2p/quic/quictransportchannel.h"
(...skipping 2035 matching lines...) Expand 10 before | Expand all | Expand 10 after
2079 } 2079 }
2080 2080
2081 void WebRtcSession::OnDtlsHandshakeError(rtc::SSLHandshakeError error) { 2081 void WebRtcSession::OnDtlsHandshakeError(rtc::SSLHandshakeError error) {
2082 if (metrics_observer_) { 2082 if (metrics_observer_) {
2083 metrics_observer_->IncrementEnumCounter( 2083 metrics_observer_->IncrementEnumCounter(
2084 webrtc::kEnumCounterDtlsHandshakeError, static_cast<int>(error), 2084 webrtc::kEnumCounterDtlsHandshakeError, static_cast<int>(error),
2085 static_cast<int>(rtc::SSLHandshakeError::MAX_VALUE)); 2085 static_cast<int>(rtc::SSLHandshakeError::MAX_VALUE));
2086 } 2086 }
2087 } 2087 }
2088 } // namespace webrtc 2088 } // namespace webrtc
OLDNEW
« no previous file with comments | « webrtc/api/peerconnection.cc ('k') | webrtc/audio/BUILD.gn » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698