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diff --git a/webrtc/video/call_stats.h b/webrtc/video/call_stats.h
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+/*
+ *  Copyright (c) 2012 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.
+ */
+
+#ifndef WEBRTC_VIDEO_CALL_STATS_H_
+#define WEBRTC_VIDEO_CALL_STATS_H_
+
+#include <list>
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/system_wrappers/include/clock.h"
+
+namespace webrtc {
+
+class CallStatsObserver;
+class CriticalSectionWrapper;
+class RtcpRttStats;
+
+// CallStats keeps track of statistics for a call.
+class CallStats : public Module {
+ public:
+  friend class RtcpObserver;
+
+  explicit CallStats(Clock* clock);
+  ~CallStats();
+
+  // Implements Module, to use the process thread.
+  int64_t TimeUntilNextProcess() override;
+  int32_t Process() override;
+
+  // Returns a RtcpRttStats to register at a statistics provider. The object
+  // has the same lifetime as the CallStats instance.
+  RtcpRttStats* rtcp_rtt_stats() const;
+
+  // Registers/deregisters a new observer to receive statistics updates.
+  void RegisterStatsObserver(CallStatsObserver* observer);
+  void DeregisterStatsObserver(CallStatsObserver* observer);
+
+  // Helper struct keeping track of the time a rtt value is reported.
+  struct RttTime {
+    RttTime(int64_t new_rtt, int64_t rtt_time)
+        : rtt(new_rtt), time(rtt_time) {}
+    const int64_t rtt;
+    const int64_t time;
+  };
+
+ protected:
+  void OnRttUpdate(int64_t rtt);
+
+  int64_t avg_rtt_ms() const;
+
+ private:
+  Clock* const clock_;
+  // Protecting all members.
+  rtc::scoped_ptr<CriticalSectionWrapper> crit_;
+  // Observer receiving statistics updates.
+  rtc::scoped_ptr<RtcpRttStats> rtcp_rtt_stats_;
+  // The last time 'Process' resulted in statistic update.
+  int64_t last_process_time_;
+  // The last RTT in the statistics update (zero if there is no valid estimate).
+  int64_t max_rtt_ms_;
+  int64_t avg_rtt_ms_;
+
+  // All Rtt reports within valid time interval, oldest first.
+  std::list<RttTime> reports_;
+
+  // Observers getting stats reports.
+  std::list<CallStatsObserver*> observers_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(CallStats);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_VIDEO_CALL_STATS_H_