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

webrtc/modules/rtp_rtcp/source/rtp_sender.h

 /*
  *  Copyright (c) 2012 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.
  */
 
@@ skipping 79 matching lines @@
  public:
   RTPSender(bool audio,
             Clock* clock,
             Transport* transport,
             RtpPacketSender* paced_sender,
             TransportSequenceNumberAllocator* sequence_number_allocator,
             TransportFeedbackObserver* transport_feedback_callback,
             BitrateStatisticsObserver* bitrate_callback,
             FrameCountObserver* frame_count_observer,
             SendSideDelayObserver* send_side_delay_observer,
-            RtcEventLog* event_log);
+            RtcEventLog* event_log,
+            SendPacketObserver* send_packet_observer);
+
   virtual ~RTPSender();
 
   void ProcessBitrate();
 
   uint16_t ActualSendBitrateKbit() const override;
 
   uint32_t VideoBitrateSent() const;
   uint32_t FecOverheadRate() const;
   uint32_t NackOverheadRate() const;
 
@@ skipping 235 matching lines @@
                           size_t padding_length);
 
   void BuildRtxPacket(uint8_t* buffer, size_t* length,
                       uint8_t* buffer_rtx);
 
   bool SendPacketToNetwork(const uint8_t* packet,
                            size_t size,
                            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.
   bool FindHeaderExtensionPosition(RTPExtensionType type,
                                    const uint8_t* rtp_packet,
                                    size_t rtp_packet_length,
                                    const RTPHeader& rtp_header,
                                    size_t* position) const;
 
   void UpdateTransmissionTimeOffset(uint8_t* rtp_packet,
                                     size_t rtp_packet_length,
                                     const RTPHeader& rtp_header,
                                     int64_t time_diff_ms) const;
   void UpdateAbsoluteSendTime(uint8_t* rtp_packet,
                               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;
+
+  bool AllocateTransportSequenceNumber(int* packet_id) const;
 
   void UpdateRtpStats(const uint8_t* buffer,
                       size_t packet_length,
                       const RTPHeader& header,
                       bool is_rtx,
                       bool is_retransmit);
   bool IsFecPacket(const uint8_t* buffer, const RTPHeader& header) const;
 
   class BitrateAggregator {
    public:
@@ skipping 69 matching lines @@
   // Statistics
   rtc::CriticalSection statistics_crit_;
   SendDelayMap send_delays_ GUARDED_BY(statistics_crit_);
   FrameCounts frame_counts_ GUARDED_BY(statistics_crit_);
   StreamDataCounters rtp_stats_ GUARDED_BY(statistics_crit_);
   StreamDataCounters rtx_rtp_stats_ GUARDED_BY(statistics_crit_);
   StreamDataCountersCallback* rtp_stats_callback_ GUARDED_BY(statistics_crit_);
   FrameCountObserver* const frame_count_observer_;
   SendSideDelayObserver* const send_side_delay_observer_;
   RtcEventLog* const event_log_;
+  SendPacketObserver* const send_packet_observer_;
 
   // RTP variables
   bool start_timestamp_forced_ GUARDED_BY(send_critsect_);
   uint32_t start_timestamp_ GUARDED_BY(send_critsect_);
   SSRCDatabase* const ssrc_db_;
   uint32_t remote_ssrc_ GUARDED_BY(send_critsect_);
   bool sequence_number_forced_ GUARDED_BY(send_critsect_);
   uint16_t sequence_number_ GUARDED_BY(send_critsect_);
   uint16_t sequence_number_rtx_ GUARDED_BY(send_critsect_);
   bool ssrc_forced_ GUARDED_BY(send_critsect_);
@@ skipping 15 matching lines @@
   // that the target bitrate is still valid.
   rtc::CriticalSection target_bitrate_critsect_;
   uint32_t target_bitrate_ GUARDED_BY(target_bitrate_critsect_);
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSender);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_