Call OnSentPacket for all packets sent in the test framework.

webrtc/modules/rtp_rtcp/source/rtp_sender.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_sender.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index 342486e7d86a03e41701ea99f0bf3d77b3aa44a7..61ed512e644259c6c7c833524eb83cd48b95c1fd 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -680,13 +680,13 @@ size_t RTPSender::SendPadData(size_t bytes,
           UpdateTransportSequenceNumber(padding_packet, length, rtp_header);
     }
 
-    if (!SendPacketToNetwork(padding_packet, length, options))
-      break;
-
     if (using_transport_seq && transport_feedback_observer_) {
       transport_feedback_observer_->AddPacket(options.packet_id, length, true);
     }
 
+    if (!SendPacketToNetwork(padding_packet, length, options))
+      break;
+
     bytes_sent += padding_bytes_in_packet;
     UpdateRtpStats(padding_packet, length, rtp_header, over_rtx, false);
   }
@@ -940,14 +940,15 @@ bool RTPSender::PrepareAndSendPacket(uint8_t* buffer,
         UpdateTransportSequenceNumber(buffer_to_send_ptr, length, rtp_header);
   }
 
+  if (using_transport_seq && transport_feedback_observer_) {
+    transport_feedback_observer_->AddPacket(options.packet_id, length, true);
+  }
+
   bool ret = SendPacketToNetwork(buffer_to_send_ptr, length, options);
   if (ret) {
     CriticalSectionScoped lock(send_critsect_.get());
     media_has_been_sent_ = true;
   }
-  if (using_transport_seq && transport_feedback_observer_) {
-    transport_feedback_observer_->AddPacket(options.packet_id, length, true);
-  }
   UpdateRtpStats(buffer_to_send_ptr, length, rtp_header, send_over_rtx,
                  is_retransmit);
   return ret;