Add histogram stats for send delay for a sent video stream.

webrtc/modules/rtp_rtcp/source/rtp_sender.h

Index: webrtc/modules/rtp_rtcp/source/rtp_sender.h
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.h b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
index 8fc0696067ad073e4d3198797170fd1005129dbd..284ae12fb0098c588ffc652fcbaac5b8fc012fe6 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
@@ -96,7 +96,8 @@ class RTPSender : public RTPSenderInterface {
             TransportFeedbackObserver* transport_feedback_callback,
             BitrateStatisticsObserver* bitrate_callback,
             FrameCountObserver* frame_count_observer,
-            SendSideDelayObserver* send_side_delay_observer);
+            SendSideDelayObserver* send_side_delay_observer,
+            SendPacketObserver* send_packet_observer);
   virtual ~RTPSender();
 
   void ProcessBitrate();
@@ -353,6 +354,7 @@ class RTPSender : public RTPSenderInterface {
                            const PacketOptions& options);
 
   void UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms);
+  void UpdateOnSendPacket(int64_t capture_time_ms, int* packet_id);
 
   // Find the byte position of the RTP extension as indicated by |type| in
   // |rtp_packet|. Return false if such extension doesn't exist.
@@ -432,6 +434,7 @@ class RTPSender : public RTPSenderInterface {
   StreamDataCountersCallback* rtp_stats_callback_ GUARDED_BY(statistics_crit_);
   FrameCountObserver* const frame_count_observer_;
   SendSideDelayObserver* const send_side_delay_observer_;
+  SendPacketObserver* const send_packet_observer_;
 
   // RTP variables
   bool start_timestamp_forced_ GUARDED_BY(send_critsect_);