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

webrtc/audio/audio_send_stream.h

 /*
  *  Copyright (c) 2015 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_AUDIO_AUDIO_SEND_STREAM_H_
 #define WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_
 
 #include <memory>
 #include <vector>
 
+#include "webrtc/audio/time_interval.h"
 #include "webrtc/call/audio_send_stream.h"
 #include "webrtc/call/audio_state.h"
 #include "webrtc/call/bitrate_allocator.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/rtc_base/constructormagic.h"
 #include "webrtc/rtc_base/thread_checker.h"
 #include "webrtc/voice_engine/transport_feedback_packet_loss_tracker.h"
 
 namespace webrtc {
 class VoiceEngine;
@@ skipping 42 matching lines @@
 
   // From PacketFeedbackObserver.
   void OnPacketAdded(uint32_t ssrc, uint16_t seq_num) override;
   void OnPacketFeedbackVector(
       const std::vector<PacketFeedback>& packet_feedback_vector) override;
 
   const webrtc::AudioSendStream::Config& config() const;
   void SetTransportOverhead(int transport_overhead_per_packet);
 
   RtpState GetRtpState() const;
+  const TimeInterval& GetActiveLifetime() const;
 
  private:
+  class TimedTransport;
+
   VoiceEngine* voice_engine() const;
 
   // These are all static to make it less likely that (the old) config_ is
   // accessed unintentionally.
   static void ConfigureStream(AudioSendStream* stream,
                               const Config& new_config,
                               bool first_time);
   static bool SetupSendCodec(AudioSendStream* stream, const Config& new_config);
   static bool ReconfigureSendCodec(AudioSendStream* stream,
                                    const Config& new_config);
@@ skipping 19 matching lines @@
   RtpTransportControllerSendInterface* const transport_;
   std::unique_ptr<RtcpBandwidthObserver> bandwidth_observer_;
 
   rtc::CriticalSection packet_loss_tracker_cs_;
   TransportFeedbackPacketLossTracker packet_loss_tracker_
       GUARDED_BY(&packet_loss_tracker_cs_);
 
   RtpRtcp* rtp_rtcp_module_;
   rtc::Optional<RtpState> const suspended_rtp_state_;
 
+  std::unique_ptr<TimedTransport> timed_send_transport_adapter_;
+  TimeInterval active_lifetime_;
+
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream);
 };
 }  // namespace internal
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_