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..2bd8c1af0b8013fe24bb3888c099e0603a082956 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 +354,9 @@ class RTPSender : public RTPSenderInterface {
                            const PacketOptions& options);
 
   void UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms);
+  void UpdateOnSendPacket(int packet_id,
+                          int64_t capture_time_ms,
+                          uint32_t ssrc);
 
   // Find the byte position of the RTP extension as indicated by |type| in
   // |rtp_packet|. Return false if such extension doesn't exist.
@@ -370,12 +374,16 @@ class RTPSender : public RTPSenderInterface {
                               size_t rtp_packet_length,
                               const RTPHeader& rtp_header,
                               int64_t now_ms) const;
-  // Update the transport sequence number of the packet using a new sequence
-  // number allocated by SequenceNumberAllocator. Returns the assigned sequence
-  // number, or 0 if extension could not be updated.
-  uint16_t UpdateTransportSequenceNumber(uint8_t* rtp_packet,
-                                         size_t rtp_packet_length,
-                                         const RTPHeader& rtp_header) const;
+
+  bool UpdateTransportSequenceNumber(uint16_t sequence_number,
+                                     uint8_t* rtp_packet,
+                                     size_t rtp_packet_length,
+                                     const RTPHeader& rtp_header) const;
+
+  // Returns true if transport sequence number is used, false otherwise.
+  // The |packet_id| is always updated with a new sequence number allocated by
+  // the SequenceNumberAllocator (if allocator exists).
+  bool UseTransportSequenceNumber(int* packet_id) const;
 
   void UpdateRtpStats(const uint8_t* buffer,
                       size_t packet_length,
@@ -432,6 +440,7 @@ 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_);