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

webrtc/call/call.cc

Index: webrtc/call/call.cc
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index 91c27c80ad99f16805b80ad45b2916c60812748d..de660b9f62eaa8a5f95dcd4f2fbf49e71efd785c 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -627,6 +627,13 @@ void Call::OnSentPacket(const rtc::SentPacket& sent_packet) {
   if (first_packet_sent_ms_ == -1)
     first_packet_sent_ms_ = clock_->TimeInMilliseconds();
   congestion_controller_->OnSentPacket(sent_packet);
+
+  ReadLockScoped read_lock(*send_crit_);
+  for (VideoSendStream* stream : video_send_streams_) {
+    // TODO(asapersson): Use sent_packet.send_time_ms.
+    if (stream->OnSentPacket(sent_packet.packet_id))
+      return;
+  }
 }
 
 void Call::OnNetworkChanged(uint32_t target_bitrate_bps, uint8_t fraction_loss,