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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 | 10 |
11 #include "webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h" | 11 #include "webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h" |
12 | 12 |
13 #include <math.h> | 13 #include <math.h> |
14 | 14 |
15 #include <cstdlib> | 15 #include <cstdlib> |
16 | 16 |
17 #include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h" | |
17 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h" | 18 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h" |
18 #include "webrtc/modules/rtp_rtcp/source/time_util.h" | 19 #include "webrtc/modules/rtp_rtcp/source/time_util.h" |
19 | 20 |
20 namespace webrtc { | 21 namespace webrtc { |
21 | 22 |
22 const int64_t kStatisticsTimeoutMs = 8000; | 23 const int64_t kStatisticsTimeoutMs = 8000; |
23 const int64_t kStatisticsProcessIntervalMs = 1000; | 24 const int64_t kStatisticsProcessIntervalMs = 1000; |
24 | 25 |
25 StreamStatistician::~StreamStatistician() {} | 26 StreamStatistician::~StreamStatistician() {} |
26 | 27 |
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269 | 270 |
270 // Store this report. | 271 // Store this report. |
271 last_reported_statistics_ = stats; | 272 last_reported_statistics_ = stats; |
272 | 273 |
273 // Only for report blocks in RTCP SR and RR. | 274 // Only for report blocks in RTCP SR and RR. |
274 last_report_inorder_packets_ = | 275 last_report_inorder_packets_ = |
275 receive_counters_.transmitted.packets - | 276 receive_counters_.transmitted.packets - |
276 receive_counters_.retransmitted.packets; | 277 receive_counters_.retransmitted.packets; |
277 last_report_old_packets_ = receive_counters_.retransmitted.packets; | 278 last_report_old_packets_ = receive_counters_.retransmitted.packets; |
278 last_report_seq_max_ = received_seq_max_; | 279 last_report_seq_max_ = received_seq_max_; |
280 BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "cumulative_loss[pkts]", \ | |
281 clock_->TimeInMilliseconds(), cumulative_loss_, ssrc_); | |
282 BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "received_seq_max[pkts]", \ | |
283 clock_->TimeInMilliseconds(), \ | |
284 (received_seq_max_ - received_seq_first_), ssrc_); | |
stefan-webrtc
2016/09/02 11:13:04
Remove all \ as they aren't needed.
| |
279 | 285 |
280 return stats; | 286 return stats; |
281 } | 287 } |
282 | 288 |
283 void StreamStatisticianImpl::GetDataCounters( | 289 void StreamStatisticianImpl::GetDataCounters( |
284 size_t* bytes_received, uint32_t* packets_received) const { | 290 size_t* bytes_received, uint32_t* packets_received) const { |
285 rtc::CritScope cs(&stream_lock_); | 291 rtc::CritScope cs(&stream_lock_); |
286 if (bytes_received) { | 292 if (bytes_received) { |
287 *bytes_received = receive_counters_.transmitted.payload_bytes + | 293 *bytes_received = receive_counters_.transmitted.payload_bytes + |
288 receive_counters_.transmitted.header_bytes + | 294 receive_counters_.transmitted.header_bytes + |
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503 void NullReceiveStatistics::SetMaxReorderingThreshold( | 509 void NullReceiveStatistics::SetMaxReorderingThreshold( |
504 int max_reordering_threshold) {} | 510 int max_reordering_threshold) {} |
505 | 511 |
506 void NullReceiveStatistics::RegisterRtcpStatisticsCallback( | 512 void NullReceiveStatistics::RegisterRtcpStatisticsCallback( |
507 RtcpStatisticsCallback* callback) {} | 513 RtcpStatisticsCallback* callback) {} |
508 | 514 |
509 void NullReceiveStatistics::RegisterRtpStatisticsCallback( | 515 void NullReceiveStatistics::RegisterRtpStatisticsCallback( |
510 StreamDataCountersCallback* callback) {} | 516 StreamDataCountersCallback* callback) {} |
511 | 517 |
512 } // namespace webrtc | 518 } // namespace webrtc |
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