Add a histogram metric tracking for how long audio RTP packets are sent

webrtc/audio/time_interval_unittest.cc

Index: webrtc/audio/time_interval_unittest.cc
diff --git a/webrtc/audio/time_interval_unittest.cc b/webrtc/audio/time_interval_unittest.cc
new file mode 100644
index 0000000000000000000000000000000000000000..592f9fc3d4aa6edd4d8fdcc328d0d491567d9f26
--- /dev/null
+++ b/webrtc/audio/time_interval_unittest.cc
@@ -0,0 +1,48 @@
+/*
+ *  Copyright 2017 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/audio/time_interval.h"
+#include "webrtc/rtc_base/fakeclock.h"
+#include "webrtc/rtc_base/timedelta.h"
+#include "webrtc/test/gtest.h"
+
+namespace webrtc {
+
+TEST(TimeIntervalTest, TimeInMs) {
+  rtc::ScopedFakeClock fake_clock;
+  TimeInterval interval;
+  interval.Extend();
+  fake_clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(100));
+  interval.Extend();
+  EXPECT_EQ(interval.Length(), 100);
+}
+
+TEST(TimeIntervalTest, Empty) {
+  TimeInterval interval;
+  EXPECT_TRUE(interval.Empty());
+  interval.Extend();
+  EXPECT_FALSE(interval.Empty());
+  interval.Extend(200);
+  EXPECT_FALSE(interval.Empty());
+}
+
+TEST(TimeIntervalTest, MonotoneIncreasing) {
+  const size_t point_count = 7;
+  const int64_t interval_points[] = {3, 2, 5, 0, 4, 1, 6};
+  const int64_t interval_differences[] = {0, 1, 3, 5, 5, 5, 6};
+  TimeInterval interval;
+  EXPECT_TRUE(interval.Empty());
+  for (size_t i = 0; i < point_count; ++i) {
+    interval.Extend(interval_points[i]);
+    EXPECT_EQ(interval_differences[i], interval.Length());
+  }
+}
+
+}  // namespace webrtc