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

webrtc/video/vie_channel.cc

Index: webrtc/video/vie_channel.cc
diff --git a/webrtc/video/vie_channel.cc b/webrtc/video/vie_channel.cc
index 18422fe3087743bf49f2a0fb658627ff2186b024..e05135305cd9f9bf3cce2ef7996483b5e400dd0f 100644
--- a/webrtc/video/vie_channel.cc
+++ b/webrtc/video/vie_channel.cc
@@ -120,6 +120,7 @@ ViEChannel::ViEChannel(Transport* transport,
                                &send_bitrate_observer_,
                                &send_frame_count_observer_,
                                &send_side_delay_observer_,
+                               &send_packet_observer_,
                                max_rtp_streams)) {
   vie_receiver_.Init(rtp_rtcp_modules_);
   if (sender_) {
@@ -287,6 +288,10 @@ void ViEChannel::RegisterSendSideDelayObserver(
   send_side_delay_observer_.Set(observer);
 }
 
+void ViEChannel::RegisterSendPacketObserver(SendPacketObserver* observer) {
+  send_packet_observer_.Set(observer);
+}
+
 void ViEChannel::RegisterSendBitrateObserver(
     BitrateStatisticsObserver* observer) {
   send_bitrate_observer_.Set(observer);
@@ -435,6 +440,7 @@ std::vector<RtpRtcp*> ViEChannel::CreateRtpRtcpModules(
     BitrateStatisticsObserver* send_bitrate_observer,
     FrameCountObserver* send_frame_count_observer,
     SendSideDelayObserver* send_side_delay_observer,
+    SendPacketObserver* send_packet_observer,
     size_t num_modules) {
   RTC_DCHECK_GT(num_modules, 0u);
   RtpRtcp::Configuration configuration;
@@ -453,6 +459,7 @@ std::vector<RtpRtcp*> ViEChannel::CreateRtpRtcpModules(
   configuration.send_bitrate_observer = send_bitrate_observer;
   configuration.send_frame_count_observer = send_frame_count_observer;
   configuration.send_side_delay_observer = send_side_delay_observer;
+  configuration.send_packet_observer = send_packet_observer;
   configuration.bandwidth_callback = bandwidth_callback;
   configuration.transport_feedback_callback = transport_feedback_callback;