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

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc

 /*
  *  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 42 matching lines @@
       bandwidth_callback(nullptr),
       transport_feedback_callback(nullptr),
       rtt_stats(nullptr),
       rtcp_packet_type_counter_observer(nullptr),
       remote_bitrate_estimator(nullptr),
       paced_sender(nullptr),
       transport_sequence_number_allocator(nullptr),
       send_bitrate_observer(nullptr),
       send_frame_count_observer(nullptr),
       send_side_delay_observer(nullptr),
-      event_log(nullptr) {}
+      event_log(nullptr),
+      send_packet_observer(nullptr) {}
 
 RtpRtcp* RtpRtcp::CreateRtpRtcp(const RtpRtcp::Configuration& configuration) {
   if (configuration.clock) {
     return new ModuleRtpRtcpImpl(configuration);
   } else {
     // No clock implementation provided, use default clock.
     RtpRtcp::Configuration configuration_copy;
     memcpy(&configuration_copy, &configuration,
            sizeof(RtpRtcp::Configuration));
     configuration_copy.clock = Clock::GetRealTimeClock();
     return new ModuleRtpRtcpImpl(configuration_copy);
   }
 }
 
 ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
     : rtp_sender_(configuration.audio,
                   configuration.clock,
                   configuration.outgoing_transport,
                   configuration.paced_sender,
                   configuration.transport_sequence_number_allocator,
                   configuration.transport_feedback_callback,
                   configuration.send_bitrate_observer,
                   configuration.send_frame_count_observer,
                   configuration.send_side_delay_observer,
-                  configuration.event_log),
+                  configuration.event_log,
+                  configuration.send_packet_observer),
       rtcp_sender_(configuration.audio,
                    configuration.clock,
                    configuration.receive_statistics,
                    configuration.rtcp_packet_type_counter_observer,
                    configuration.event_log,
                    configuration.outgoing_transport),
       rtcp_receiver_(configuration.clock,
                      configuration.receiver_only,
                      configuration.rtcp_packet_type_counter_observer,
                      configuration.bandwidth_callback,
                      configuration.intra_frame_callback,
                      configuration.transport_feedback_callback,
                      this),
       clock_(configuration.clock),
       audio_(configuration.audio),
       collision_detected_(false),
       last_process_time_(configuration.clock->TimeInMilliseconds()),
       last_bitrate_process_time_(configuration.clock->TimeInMilliseconds()),
       last_rtt_process_time_(configuration.clock->TimeInMilliseconds()),
-      packet_overhead_(28),                     // IPV4 UDP.
+      packet_overhead_(28),  // IPV4 UDP.
       nack_last_time_sent_full_(0),
       nack_last_time_sent_full_prev_(0),
       nack_last_seq_number_sent_(0),
       key_frame_req_method_(kKeyFrameReqPliRtcp),
       remote_bitrate_(configuration.remote_bitrate_estimator),
       rtt_stats_(configuration.rtt_stats),
       rtt_ms_(0) {
   // Make sure that RTCP objects are aware of our SSRC.
   uint32_t SSRC = rtp_sender_.SSRC();
   rtcp_sender_.SetSSRC(SSRC);
@@ skipping 865 matching lines @@
 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback(
     StreamDataCountersCallback* callback) {
   rtp_sender_.RegisterRtpStatisticsCallback(callback);
 }
 
 StreamDataCountersCallback*
     ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
   return rtp_sender_.GetRtpStatisticsCallback();
 }
 }  // namespace webrtc