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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() { |
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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_ |
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