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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/vie_receiver.h" | 11 #include "webrtc/video/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/config.h" | 16 #include "webrtc/config.h" |
17 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimat
or.h" | 17 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimat
or.h" |
18 #include "webrtc/modules/rtp_rtcp/include/fec_receiver.h" | 18 #include "webrtc/modules/rtp_rtcp/include/fec_receiver.h" |
19 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h" | 19 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h" |
20 #include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h" | 20 #include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h" |
21 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" | 21 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" |
22 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h" | 22 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h" |
23 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" | 23 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" |
24 #include "webrtc/modules/video_coding/include/video_coding.h" | 24 #include "webrtc/modules/video_coding/include/video_coding.h" |
25 #include "webrtc/system_wrappers/include/metrics.h" | 25 #include "webrtc/system_wrappers/include/metrics.h" |
26 #include "webrtc/system_wrappers/include/tick_util.h" | |
27 #include "webrtc/system_wrappers/include/timestamp_extrapolator.h" | 26 #include "webrtc/system_wrappers/include/timestamp_extrapolator.h" |
28 #include "webrtc/system_wrappers/include/trace.h" | 27 #include "webrtc/system_wrappers/include/trace.h" |
29 | 28 |
30 namespace webrtc { | 29 namespace webrtc { |
31 | 30 |
32 static const int kPacketLogIntervalMs = 10000; | 31 static const int kPacketLogIntervalMs = 10000; |
33 | 32 |
34 ViEReceiver::ViEReceiver(VideoCodingModule* module_vcm, | 33 ViEReceiver::ViEReceiver(VideoCodingModule* module_vcm, |
35 RemoteBitrateEstimator* remote_bitrate_estimator, | 34 RemoteBitrateEstimator* remote_bitrate_estimator, |
36 RtpFeedback* rtp_feedback) | 35 RtpFeedback* rtp_feedback) |
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380 rtp_receive_statistics_->GetStatistician(header.ssrc); | 379 rtp_receive_statistics_->GetStatistician(header.ssrc); |
381 if (!statistician) | 380 if (!statistician) |
382 return false; | 381 return false; |
383 // Check if this is a retransmission. | 382 // Check if this is a retransmission. |
384 int64_t min_rtt = 0; | 383 int64_t min_rtt = 0; |
385 rtp_rtcp_[0]->RTT(rtp_receiver_->SSRC(), nullptr, nullptr, &min_rtt, nullptr); | 384 rtp_rtcp_[0]->RTT(rtp_receiver_->SSRC(), nullptr, nullptr, &min_rtt, nullptr); |
386 return !in_order && | 385 return !in_order && |
387 statistician->IsRetransmitOfOldPacket(header, min_rtt); | 386 statistician->IsRetransmitOfOldPacket(header, min_rtt); |
388 } | 387 } |
389 } // namespace webrtc | 388 } // namespace webrtc |
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