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

webrtc/video/vie_channel.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 75 matching lines @@
                          int payload_type_fec);
 
   RtpState GetRtpStateForSsrc(uint32_t ssrc) const;
 
   // Gets send statistics for the rtp and rtx stream.
   void GetSendStreamDataCounters(StreamDataCounters* rtp_counters,
                                  StreamDataCounters* rtx_counters) const;
 
   void RegisterSendSideDelayObserver(SendSideDelayObserver* observer);
 
+  void RegisterSendPacketObserver(SendPacketObserver* observer);
+
   // Called on any new send bitrate estimate.
   void RegisterSendBitrateObserver(BitrateStatisticsObserver* observer);
 
   // Implements RtpFeedback.
   int32_t OnInitializeDecoder(const int8_t payload_type,
                               const char payload_name[RTP_PAYLOAD_NAME_SIZE],
                               const int frequency,
                               const size_t channels,
                               const uint32_t rate) override;
   void OnIncomingSSRCChanged(const uint32_t ssrc) override;
@@ skipping 70 matching lines @@
       RtcpBandwidthObserver* bandwidth_callback,
       TransportFeedbackObserver* transport_feedback_callback,
       RtcpRttStats* rtt_stats,
       RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer,
       RemoteBitrateEstimator* remote_bitrate_estimator,
       RtpPacketSender* paced_sender,
       TransportSequenceNumberAllocator* transport_sequence_number_allocator,
       BitrateStatisticsObserver* send_bitrate_observer,
       FrameCountObserver* send_frame_count_observer,
       SendSideDelayObserver* send_side_delay_observer,
+      SendPacketObserver* send_packet_observer,
       size_t num_modules);
 
   // Assumed to be protected.
   void StartDecodeThread();
   void StopDecodeThread();
 
   void ProcessNACKRequest(const bool enable);
   // Compute NACK list parameters for the buffering mode.
   int GetRequiredNackListSize(int target_delay_ms);
 
@@ skipping 51 matching lines @@
       public RegisterableCallback<SendSideDelayObserver> {
     void SendSideDelayUpdated(int avg_delay_ms,
                               int max_delay_ms,
                               uint32_t ssrc) override {
       rtc::CritScope lock(&critsect_);
       if (callback_)
         callback_->SendSideDelayUpdated(avg_delay_ms, max_delay_ms, ssrc);
     }
   } send_side_delay_observer_;
 
+  class RegisterableSendPacketObserver
+      : public RegisterableCallback<SendPacketObserver> {
+    void OnSendPacket(uint16_t packet_id,
+                      int64_t capture_time_ms,
+                      uint32_t ssrc) override {
+      rtc::CritScope lock(&critsect_);
+      if (callback_)
+        callback_->OnSendPacket(packet_id, capture_time_ms, ssrc);
+    }
+  } send_packet_observer_;
+
   class RegisterableRtcpPacketTypeCounterObserver
       : public RegisterableCallback<RtcpPacketTypeCounterObserver> {
    public:
     void RtcpPacketTypesCounterUpdated(
         uint32_t ssrc,
         const RtcpPacketTypeCounter& packet_counter) override {
       rtc::CritScope lock(&critsect_);
       if (callback_)
         callback_->RtcpPacketTypesCounterUpdated(ssrc, packet_counter);
     }
@@ skipping 35 matching lines @@
 
   int64_t last_rtt_ms_ GUARDED_BY(crit_);
 
   // RtpRtcp modules, declared last as they use other members on construction.
   const std::vector<RtpRtcp*> rtp_rtcp_modules_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_VIE_CHANNEL_H_