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

webrtc/audio/audio_send_stream.h

Index: webrtc/audio/audio_send_stream.h
diff --git a/webrtc/audio/audio_send_stream.h b/webrtc/audio/audio_send_stream.h
index 70cf1c874d46b41651b67451bad58227ce6e400a..42a04aee0932b3d06ea0508d1b40373934c02c73 100644
--- a/webrtc/audio/audio_send_stream.h
+++ b/webrtc/audio/audio_send_stream.h
@@ -14,6 +14,7 @@
 #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"
@@ -76,8 +77,11 @@ class AudioSendStream final : public webrtc::AudioSendStream,
   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
@@ -117,6 +121,9 @@ class AudioSendStream final : public webrtc::AudioSendStream,
   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