[rtp_rtcp] ReportBlockInformation cleaned and moved out of the rtcp_receiver_help.

webrtc/modules/rtp_rtcp/source/report_block_information.cc

+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/report_block_information.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/trace_event.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
+#include "webrtc/modules/rtp_rtcp/source/time_util.h"
+
+namespace webrtc {
+namespace rtcp {
+
+ReportBlockInformation::ReportBlockInformation()
+    : number_rtt_(0), last_rtt_(0), min_rtt_(0), max_rtt_(0), sum_rtt_(0) {
+  memset(&last_received_block_, 0, sizeof(last_received_block_));
+}
+
+uint32_t ReportBlockInformation::LastRttMs() const {
+  return CompactNtpIntervalToMs(last_rtt_);
+}
+uint32_t ReportBlockInformation::MinRttMs() const {
+  return CompactNtpIntervalToMs(min_rtt_);
+}
+uint32_t ReportBlockInformation::MaxRttMs() const {
+  return CompactNtpIntervalToMs(max_rtt_);
+}
+uint32_t ReportBlockInformation::AvgRttMs() const {
+  RTC_CHECK_GT(number_rtt_, 0u);

[philipel, 2016/02/29 13:21:47]

Expected value first:
RTC_CHECK_GT(0u, number_rtt_);

[danilchap, 2016/02/29 14:41:27]

Disagree,
Code search shows that expected value last is used for CHECK_GT/LT/LE/GE
There are reasons to use expected value last even for CHECK_EQ.

[philipel, 2016/03/03 14:00:58]

I think you are right. Shouldn't it be DCHECK instead of CHECK though?

[danilchap, 2016/03/03 14:44:48]

Done.

+  return CompactNtpIntervalToMs(sum_rtt_ / number_rtt_);
+}
+
+void ReportBlockInformation::AddBlock(const rtcp::ReportBlock& block,
+                                      uint32_t remote_ssrc,
+                                      NtpTime now) {
+  last_received_block_.remoteSSRC = remote_ssrc;
+  last_received_block_.sourceSSRC = block.source_ssrc();
+  last_received_block_.fractionLost = block.fraction_lost();
+  last_received_block_.cumulativeLost = block.cumulative_lost();
+  last_received_block_.extendedHighSeqNum = block.extended_high_seq_num();
+  last_received_block_.jitter = block.jitter();
+  last_received_block_.lastSR = block.last_sr();
+  last_received_block_.delaySinceLastSR = block.delay_since_last_sr();
+
+  AddRtt(CompactNtp(now));
+}
+
+void ReportBlockInformation::AddBlock(
+    const RTCPUtility::RTCPPacketReportBlockItem& block,
+    uint32_t remote_ssrc,
+    NtpTime now) {
+  last_received_block_.remoteSSRC = remote_ssrc;
+  last_received_block_.sourceSSRC = block.SSRC;
+  last_received_block_.fractionLost = block.FractionLost;
+  last_received_block_.cumulativeLost = block.CumulativeNumOfPacketsLost;
+  last_received_block_.extendedHighSeqNum = block.ExtendedHighestSequenceNumber;
+  last_received_block_.jitter = block.Jitter;
+  last_received_block_.lastSR = block.LastSR;
+  last_received_block_.delaySinceLastSR = block.DelayLastSR;
+
+  AddRtt(CompactNtp(now));
+}
+
+void ReportBlockInformation::AddRtt(uint32_t receive_time) {
+  uint32_t send_time = last_received_block_.lastSR;
+  uint32_t rtt_ms = 0;
+  if (send_time > 0) {
+    uint32_t delay = last_received_block_.delaySinceLastSR;
+
+    // RTT in 1/(2^16) seconds.
+    uint32_t rtt_ntp = receive_time - delay - send_time;
+
+    last_rtt_ = rtt_ntp;
+    if (number_rtt_ == 0) {
+      min_rtt_ = rtt_ntp;
+      max_rtt_ = rtt_ntp;
+    } else {
+      if (min_rtt_ > rtt_ntp)
+        min_rtt_ = rtt_ntp;
+      if (max_rtt_ < rtt_ntp)
+        max_rtt_ = rtt_ntp;
+    }
+    sum_rtt_ += rtt_ntp;
+    ++number_rtt_;
+    rtt_ms = CompactNtpIntervalToMs(last_rtt_);
+  }
+  TRACE_COUNTER_ID1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "RR_RTT",
+                    last_received_block_.sourceSSRC, rtt_ms);
+}
+}  // namespace rtcp
+}  // namespace webrtc