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

webrtc/rtc_base/time_interval.h

+/*
+ *  Copyright (c) 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.
+ */
+
+#ifndef WEBRTC_RTC_BASE_TIME_INTERVAL_H_

[ossu, 2017/07/14 11:13:15]

I'm not sure if this class belongs in rtc_base just yet. Since it's got a quite
narrow scope and is only used by Call and AudioSendStream, maybe put it in call/
for now? If it's needed elsewhere later, it can be moved to rtc_base then,
possibly being adapted for a wider scope of use.

[saza WebRTC, 2017/07/17 14:27:29]

Done.

+#define WEBRTC_RTC_BASE_TIME_INTERVAL_H_
+
+#include <stdint.h>
+
+#include "webrtc/rtc_base/optional.h"
+
+namespace rtc {
+
+// This class logs the first and last time its Extend() function is called.
+// Extend(...) takes arbitrary times.
+//
+// This class is not thread-safe; Extend() should only be used when only one
+// thread may access it, such as within a lock or destructor.
+//
+// Example usage:
+//   // let x < y < z < u < v
+//   rtc::TimeInterval interval;
+//   ...  //   interval.Extend(); // at time x
+//   ...
+//   interval.Extend(); // at time y
+//   ...
+//   interval.Extend(); // at time u
+//   ...
+//   interval.Extend(z); // at time v
+//   ...
+//   rtc::Optional<int64_t> first = interval.First();
+//   rtc::Optional<int64_t> last = interval.Last();
+//   if (first) {
+//     int64_t active_time = *last - *first; // returns (u - x)
+//   }
+class TimeInterval {
+ public:
+  TimeInterval();
+  ~TimeInterval();
+  // Extend the interval with the current time.
+  void Extend();
+  // Extend the interval with a given time.
+  void Extend(int64_t time);

[pbos-webrtc, 2017/07/14 17:04:01]

Should Extend not be able to take another time interval? Then you can do:

sent_rtp_audio_timer_ms_.Extend(*audio_send_stream->GetActiveLifetime());

In call.cc, without checking empty, etc. That would just be done inside Extend.

[saza WebRTC, 2017/07/17 14:27:29]

Done.

+  // True iff Extend has never been called.
+  bool Empty() const;
+  // Returns the time of the first tick in milliseconds.
+  Optional<int64_t> First() const;
+  // Returns the time of the last tick in milliseconds.
+  Optional<int64_t> Last() const;
+
+ private:
+  Optional<int64_t> first_;
+  Optional<int64_t> last_;
+};
+
+}  // namespace rtc
+
+#endif  // WEBRTC_RTC_BASE_TIME_INTERVAL_H_