1 files changed, 44 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/include/report_block_data.cc b/third_party/libwebrtc/modules/rtp_rtcp/include/report_block_data.cc
new file mode 100644
index 0000000000..ec4d9d82e0
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/include/report_block_data.cc
@@ -0,0 +1,44 @@
+/*
+ *  Copyright 2019 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 "modules/rtp_rtcp/include/report_block_data.h"
+
+namespace webrtc {
+
+ReportBlockData::ReportBlockData()
+    : report_block_(),
+      report_block_timestamp_utc_us_(0),
+      last_rtt_ms_(0),
+      min_rtt_ms_(0),
+      max_rtt_ms_(0),
+      sum_rtt_ms_(0),
+      num_rtts_(0) {}
+
+double ReportBlockData::AvgRttMs() const {
+  return num_rtts_ ? static_cast<double>(sum_rtt_ms_) / num_rtts_ : 0.0;
+}
+
+void ReportBlockData::SetReportBlock(RTCPReportBlock report_block,
+                                     int64_t report_block_timestamp_utc_us) {
+  report_block_ = report_block;
+  report_block_timestamp_utc_us_ = report_block_timestamp_utc_us;
+}
+
+void ReportBlockData::AddRoundTripTimeSample(int64_t rtt_ms) {
+  if (rtt_ms > max_rtt_ms_)
+    max_rtt_ms_ = rtt_ms;
+  if (num_rtts_ == 0 || rtt_ms < min_rtt_ms_)
+    min_rtt_ms_ = rtt_ms;
+  last_rtt_ms_ = rtt_ms;
+  sum_rtt_ms_ += rtt_ms;
+  ++num_rtts_;
+}
+
+}  // namespace webrtc