Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(181)

Side by Side Diff: webrtc/test/rtp_rtcp_observer.h

Issue 1369263002: Unify Transport and newapi::Transport interfaces. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: self-review Created 5 years, 2 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/test/null_transport.h ('k') | webrtc/transport.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 (c) 2013 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2013 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 #ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_RTP_RTCP_OBSERVER_H_ 10 #ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_RTP_RTCP_OBSERVER_H_
11 #define WEBRTC_VIDEO_ENGINE_TEST_COMMON_RTP_RTCP_OBSERVER_H_ 11 #define WEBRTC_VIDEO_ENGINE_TEST_COMMON_RTP_RTCP_OBSERVER_H_
12 12
13 #include <map> 13 #include <map>
14 #include <vector> 14 #include <vector>
15 15
16 #include "testing/gtest/include/gtest/gtest.h" 16 #include "testing/gtest/include/gtest/gtest.h"
17 17
18 #include "webrtc/base/criticalsection.h" 18 #include "webrtc/base/criticalsection.h"
19 #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" 19 #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
20 #include "webrtc/test/direct_transport.h" 20 #include "webrtc/test/direct_transport.h"
21 #include "webrtc/typedefs.h" 21 #include "webrtc/typedefs.h"
22 #include "webrtc/video_send_stream.h" 22 #include "webrtc/video_send_stream.h"
23 23
24 namespace webrtc { 24 namespace webrtc {
25 namespace test { 25 namespace test {
26 26
27 class RtpRtcpObserver { 27 class RtpRtcpObserver {
28 public: 28 public:
29 virtual ~RtpRtcpObserver() {} 29 virtual ~RtpRtcpObserver() {}
30 newapi::Transport* SendTransport() { 30 Transport* SendTransport() {
31 return &send_transport_; 31 return &send_transport_;
32 } 32 }
33 33
34 newapi::Transport* ReceiveTransport() { 34 Transport* ReceiveTransport() {
35 return &receive_transport_; 35 return &receive_transport_;
36 } 36 }
37 37
38 virtual void SetReceivers(PacketReceiver* send_transport_receiver, 38 virtual void SetReceivers(PacketReceiver* send_transport_receiver,
39 PacketReceiver* receive_transport_receiver) { 39 PacketReceiver* receive_transport_receiver) {
40 send_transport_.SetReceiver(send_transport_receiver); 40 send_transport_.SetReceiver(send_transport_receiver);
41 receive_transport_.SetReceiver(receive_transport_receiver); 41 receive_transport_.SetReceiver(receive_transport_receiver);
42 } 42 }
43 43
44 void StopSending() { 44 void StopSending() {
(...skipping 128 matching lines...) Expand 10 before | Expand all | Expand 10 after
173 const rtc::scoped_ptr<RtpHeaderParser> parser_; 173 const rtc::scoped_ptr<RtpHeaderParser> parser_;
174 PacketTransport send_transport_, receive_transport_; 174 PacketTransport send_transport_, receive_transport_;
175 175
176 private: 176 private:
177 unsigned int timeout_ms_; 177 unsigned int timeout_ms_;
178 }; 178 };
179 } // namespace test 179 } // namespace test
180 } // namespace webrtc 180 } // namespace webrtc
181 181
182 #endif // WEBRTC_VIDEO_ENGINE_TEST_COMMON_RTP_RTCP_OBSERVER_H_ 182 #endif // WEBRTC_VIDEO_ENGINE_TEST_COMMON_RTP_RTCP_OBSERVER_H_
OLDNEW
« no previous file with comments | « webrtc/test/null_transport.h ('k') | webrtc/transport.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698