Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(255)

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

Issue 1739783002: Revert of Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 9 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 | « webrtc/api/remoteaudiosource.h ('k') | webrtc/audio/audio_receive_stream.cc » ('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
11 #include "webrtc/api/webrtcsession.h" 11 #include "webrtc/api/webrtcsession.h"
12 12
13 #include <limits.h> 13 #include <limits.h>
14 14
15 #include <algorithm> 15 #include <algorithm>
16 #include <set> 16 #include <set>
17 #include <utility> 17 #include <utility>
18 #include <vector> 18 #include <vector>
19 19
20 #include "webrtc/api/jsepicecandidate.h" 20 #include "webrtc/api/jsepicecandidate.h"
21 #include "webrtc/api/jsepsessiondescription.h" 21 #include "webrtc/api/jsepsessiondescription.h"
22 #include "webrtc/api/mediaconstraintsinterface.h" 22 #include "webrtc/api/mediaconstraintsinterface.h"
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/audio_sink.h" 26 #include "webrtc/audio/audio_sink.h"
27 #include "webrtc/base/basictypes.h" 27 #include "webrtc/base/basictypes.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.h"
34 #include "webrtc/media/base/constants.h" 34 #include "webrtc/media/base/constants.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"
(...skipping 2082 matching lines...) Expand 10 before | Expand all | Expand 10 after
2119 } 2119 }
2120 } 2120 }
2121 2121
2122 void WebRtcSession::OnSentPacket_w(cricket::TransportChannel* channel, 2122 void WebRtcSession::OnSentPacket_w(cricket::TransportChannel* channel,
2123 const rtc::SentPacket& sent_packet) { 2123 const rtc::SentPacket& sent_packet) {
2124 RTC_DCHECK(worker_thread()->IsCurrent()); 2124 RTC_DCHECK(worker_thread()->IsCurrent());
2125 media_controller_->call_w()->OnSentPacket(sent_packet); 2125 media_controller_->call_w()->OnSentPacket(sent_packet);
2126 } 2126 }
2127 2127
2128 } // namespace webrtc 2128 } // namespace webrtc
OLDNEW
« no previous file with comments | « webrtc/api/remoteaudiosource.h ('k') | webrtc/audio/audio_receive_stream.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698