Call OnSentPacket for all packets sent in the test framework.

webrtc/modules/rtp_rtcp/source/rtp_sender.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 662 matching lines @@
     }
 
     UpdateAbsoluteSendTime(padding_packet, length, rtp_header, now_ms);
 
     PacketOptions options;
     if (using_transport_seq) {
       options.packet_id =
           UpdateTransportSequenceNumber(padding_packet, length, rtp_header);
     }
 
+    if (using_transport_seq && transport_feedback_observer_) {
+      transport_feedback_observer_->AddPacket(options.packet_id, length, true);
+    }
+
     if (!SendPacketToNetwork(padding_packet, length, options))
       break;
 
-    if (using_transport_seq && transport_feedback_observer_) {
-      transport_feedback_observer_->AddPacket(options.packet_id, length, true);
-    }
-
     bytes_sent += padding_bytes_in_packet;
     UpdateRtpStats(padding_packet, length, rtp_header, over_rtx, false);
   }
 
   return bytes_sent;
 }
 
 void RTPSender::SetStorePacketsStatus(bool enable, uint16_t number_to_store) {
   packet_history_.SetStorePacketsStatus(enable, number_to_store);
 }
@@ skipping 233 matching lines @@
   bool using_transport_seq = rtp_header_extension_map_.IsRegistered(
                                  kRtpExtensionTransportSequenceNumber) &&
                              transport_sequence_number_allocator_;
 
   PacketOptions options;
   if (using_transport_seq) {
     options.packet_id =
         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;
 }
 
 void RTPSender::UpdateRtpStats(const uint8_t* buffer,
                                size_t packet_length,
                                const RTPHeader& header,
                                bool is_rtx,
                                bool is_retransmit) {
@@ skipping 941 matching lines @@
   CriticalSectionScoped lock(send_critsect_.get());
 
   RtpState state;
   state.sequence_number = sequence_number_rtx_;
   state.start_timestamp = start_timestamp_;
 
   return state;
 }
 
 }  // namespace webrtc