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

Issue 1368943002: Fix suspend below min bitrate in new API by making it possible to set min bitrate at the receive-si… (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Merge. Created 5 years, 2 months ago
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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_
(...skipping 153 matching lines...) Expand 10 before | Expand all | Expand 10 after
164 rtc::CriticalSection* const crit_; 164 rtc::CriticalSection* const crit_;
165 165
166 RtpRtcpObserver* const observer_; 166 RtpRtcpObserver* const observer_;
167 const PacketTransportAction on_rtp_, on_rtcp_; 167 const PacketTransportAction on_rtp_, on_rtcp_;
168 }; 168 };
169 169
170 protected: 170 protected:
171 rtc::CriticalSection crit_; 171 rtc::CriticalSection crit_;
172 const rtc::scoped_ptr<EventWrapper> observation_complete_; 172 const rtc::scoped_ptr<EventWrapper> observation_complete_;
173 const rtc::scoped_ptr<RtpHeaderParser> parser_; 173 const rtc::scoped_ptr<RtpHeaderParser> parser_;
174 PacketTransport send_transport_, receive_transport_;
174 175
175 private: 176 private:
176 PacketTransport send_transport_, receive_transport_;
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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