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Side by Side Diff: webrtc/video_engine/vie_receiver.cc

Issue 1413333002: system_wrappers: rename interface -> include (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebased again! Created 5 years, 1 month ago
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1 /* 1 /*
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 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/video_engine/vie_receiver.h" 11 #include "webrtc/video_engine/vie_receiver.h"
12 12
13 #include <vector> 13 #include <vector>
14 14
15 #include "webrtc/base/logging.h" 15 #include "webrtc/base/logging.h"
16 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimat or.h" 16 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimat or.h"
17 #include "webrtc/modules/rtp_rtcp/interface/fec_receiver.h" 17 #include "webrtc/modules/rtp_rtcp/interface/fec_receiver.h"
18 #include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h" 18 #include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
19 #include "webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h" 19 #include "webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h"
20 #include "webrtc/modules/rtp_rtcp/interface/rtp_cvo.h" 20 #include "webrtc/modules/rtp_rtcp/interface/rtp_cvo.h"
21 #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" 21 #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
22 #include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h" 22 #include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
23 #include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h" 23 #include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h"
24 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" 24 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
25 #include "webrtc/modules/video_coding/main/interface/video_coding.h" 25 #include "webrtc/modules/video_coding/main/interface/video_coding.h"
26 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" 26 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
27 #include "webrtc/system_wrappers/interface/metrics.h" 27 #include "webrtc/system_wrappers/include/metrics.h"
28 #include "webrtc/system_wrappers/interface/tick_util.h" 28 #include "webrtc/system_wrappers/include/tick_util.h"
29 #include "webrtc/system_wrappers/interface/timestamp_extrapolator.h" 29 #include "webrtc/system_wrappers/include/timestamp_extrapolator.h"
30 #include "webrtc/system_wrappers/interface/trace.h" 30 #include "webrtc/system_wrappers/include/trace.h"
31 31
32 namespace webrtc { 32 namespace webrtc {
33 33
34 static const int kPacketLogIntervalMs = 10000; 34 static const int kPacketLogIntervalMs = 10000;
35 35
36 ViEReceiver::ViEReceiver(VideoCodingModule* module_vcm, 36 ViEReceiver::ViEReceiver(VideoCodingModule* module_vcm,
37 RemoteBitrateEstimator* remote_bitrate_estimator, 37 RemoteBitrateEstimator* remote_bitrate_estimator,
38 RtpFeedback* rtp_feedback) 38 RtpFeedback* rtp_feedback)
39 : receive_cs_(CriticalSectionWrapper::CreateCriticalSection()), 39 : receive_cs_(CriticalSectionWrapper::CreateCriticalSection()),
40 clock_(Clock::GetRealTimeClock()), 40 clock_(Clock::GetRealTimeClock()),
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473 rtp_receive_statistics_->GetStatistician(header.ssrc); 473 rtp_receive_statistics_->GetStatistician(header.ssrc);
474 if (!statistician) 474 if (!statistician)
475 return false; 475 return false;
476 // Check if this is a retransmission. 476 // Check if this is a retransmission.
477 int64_t min_rtt = 0; 477 int64_t min_rtt = 0;
478 rtp_rtcp_->RTT(rtp_receiver_->SSRC(), NULL, NULL, &min_rtt, NULL); 478 rtp_rtcp_->RTT(rtp_receiver_->SSRC(), NULL, NULL, &min_rtt, NULL);
479 return !in_order && 479 return !in_order &&
480 statistician->IsRetransmitOfOldPacket(header, min_rtt); 480 statistician->IsRetransmitOfOldPacket(header, min_rtt);
481 } 481 }
482 } // namespace webrtc 482 } // namespace webrtc
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