Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(357)

Side by Side Diff: webrtc/voice_engine/channel_proxy.cc

Issue 2808043002: Move RtpTransportControllerSend to a new file. (Closed)
Patch Set: Rebase. Created 3 years, 8 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/voice_engine/channel.cc ('k') | no next file » | 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 (c) 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2015 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/voice_engine/channel_proxy.h" 11 #include "webrtc/voice_engine/channel_proxy.h"
12 12
13 #include <utility> 13 #include <utility>
14 14
15 #include "webrtc/api/call/audio_sink.h" 15 #include "webrtc/api/call/audio_sink.h"
16 #include "webrtc/base/checks.h" 16 #include "webrtc/base/checks.h"
17 #include "webrtc/base/logging.h" 17 #include "webrtc/base/logging.h"
18 #include "webrtc/call/rtp_transport_controller_send.h" 18 #include "webrtc/call/rtp_transport_controller_send_interface.h"
19 #include "webrtc/voice_engine/channel.h" 19 #include "webrtc/voice_engine/channel.h"
20 20
21 namespace webrtc { 21 namespace webrtc {
22 namespace voe { 22 namespace voe {
23 ChannelProxy::ChannelProxy() : channel_owner_(nullptr) {} 23 ChannelProxy::ChannelProxy() : channel_owner_(nullptr) {}
24 24
25 ChannelProxy::ChannelProxy(const ChannelOwner& channel_owner) : 25 ChannelProxy::ChannelProxy(const ChannelOwner& channel_owner) :
26 channel_owner_(channel_owner) { 26 channel_owner_(channel_owner) {
27 RTC_CHECK(channel_owner_.channel()); 27 RTC_CHECK(channel_owner_.channel());
28 module_process_thread_checker_.DetachFromThread(); 28 module_process_thread_checker_.DetachFromThread();
(...skipping 378 matching lines...) Expand 10 before | Expand all | Expand 10 after
407 return channel()->GetSources(); 407 return channel()->GetSources();
408 } 408 }
409 409
410 Channel* ChannelProxy::channel() const { 410 Channel* ChannelProxy::channel() const {
411 RTC_DCHECK(channel_owner_.channel()); 411 RTC_DCHECK(channel_owner_.channel());
412 return channel_owner_.channel(); 412 return channel_owner_.channel();
413 } 413 }
414 414
415 } // namespace voe 415 } // namespace voe
416 } // namespace webrtc 416 } // namespace webrtc
OLDNEW
« no previous file with comments | « webrtc/voice_engine/channel.cc ('k') | no next file » | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698