Use CallStats for RTT in Call, rather than VideoSendStream::GetRtt()

webrtc/video/call_stats_unittest.cc

 /*
  *  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.
  */
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/system_wrappers/include/metrics.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
+#include "webrtc/test/histogram.h"
 #include "webrtc/video/call_stats.h"
 
 using ::testing::_;
 using ::testing::AnyNumber;
 using ::testing::Return;
 
 namespace webrtc {
 
 class MockStatsObserver : public CallStatsObserver {
  public:
@@ skipping 11 matching lines @@
   virtual void SetUp() { call_stats_.reset(new CallStats(&fake_clock_)); }
   SimulatedClock fake_clock_;
   rtc::scoped_ptr<CallStats> call_stats_;
 };
 
 TEST_F(CallStatsTest, AddAndTriggerCallback) {
   MockStatsObserver stats_observer;
   RtcpRttStats* rtcp_rtt_stats = call_stats_->rtcp_rtt_stats();
   call_stats_->RegisterStatsObserver(&stats_observer);
   fake_clock_.AdvanceTimeMilliseconds(1000);
-  EXPECT_EQ(0, rtcp_rtt_stats->LastProcessedRtt());
+  EXPECT_EQ(-1, rtcp_rtt_stats->LastProcessedRtt());
 
   const int64_t kRtt = 25;
   rtcp_rtt_stats->OnRttUpdate(kRtt);
   EXPECT_CALL(stats_observer, OnRttUpdate(kRtt, kRtt)).Times(1);
   call_stats_->Process();
   EXPECT_EQ(kRtt, rtcp_rtt_stats->LastProcessedRtt());
 
   const int64_t kRttTimeOutMs = 1500 + 10;
   fake_clock_.AdvanceTimeMilliseconds(kRttTimeOutMs);
   EXPECT_CALL(stats_observer, OnRttUpdate(_, _)).Times(0);
   call_stats_->Process();
-  EXPECT_EQ(0, rtcp_rtt_stats->LastProcessedRtt());
+  EXPECT_EQ(-1, rtcp_rtt_stats->LastProcessedRtt());
 
   call_stats_->DeregisterStatsObserver(&stats_observer);
 }
 
 TEST_F(CallStatsTest, ProcessTime) {
   MockStatsObserver stats_observer;
   call_stats_->RegisterStatsObserver(&stats_observer);
   RtcpRttStats* rtcp_rtt_stats = call_stats_->rtcp_rtt_stats();
   rtcp_rtt_stats->OnRttUpdate(100);
 
@@ skipping 123 matching lines @@
   fake_clock_.AdvanceTimeMilliseconds(1000);
   rtcp_rtt_stats->OnRttUpdate(kRttLow);
   rtcp_rtt_stats->OnRttUpdate(kRttHigh);
   EXPECT_CALL(stats_observer, OnRttUpdate(kAvgRtt, kRttHigh)).Times(1);
   call_stats_->Process();
   EXPECT_EQ(kAvgRtt, rtcp_rtt_stats->LastProcessedRtt());
 
   call_stats_->DeregisterStatsObserver(&stats_observer);
 }
 
+TEST_F(CallStatsTest, ProducesHistogramMetrics) {
+  const int64_t kRtt = 123;
+  RtcpRttStats* rtcp_rtt_stats = call_stats_->rtcp_rtt_stats();
+  rtcp_rtt_stats->OnRttUpdate(kRtt);
+  fake_clock_.AdvanceTimeMilliseconds(metrics::kMinRunTimeInSeconds * 1000);
+  rtcp_rtt_stats->OnRttUpdate(kRtt);
+  call_stats_->Process();
+  call_stats_.reset();
+
+  EXPECT_EQ(1, test::NumHistogramSamples(
+                   "WebRTC.Video.AverageRoundTripTimeInMilliseconds"));
+  EXPECT_EQ(kRtt, test::LastHistogramSample(
+                      "WebRTC.Video.AverageRoundTripTimeInMilliseconds"));
+}
+
 }  // namespace webrtc