Remove audio/video distinction for probe packets.

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 658 matching lines @@
           padding_packet, length, rtp_header, now_ms - capture_time_ms);
     }
 
     UpdateAbsoluteSendTime(padding_packet, length, rtp_header, now_ms);
 
     PacketOptions options;
     if (AllocateTransportSequenceNumber(&options.packet_id)) {
       if (UpdateTransportSequenceNumber(options.packet_id, padding_packet,
                                         length, rtp_header)) {
         if (transport_feedback_observer_)
-          transport_feedback_observer_->AddPacket(options.packet_id, length,
-                                                  true);
+          transport_feedback_observer_->AddPacket(options.packet_id, length);
       }
     }
 
     if (!SendPacketToNetwork(padding_packet, length, options))
       break;
 
     bytes_sent += padding_bytes_in_packet;
     UpdateRtpStats(padding_packet, length, rtp_header, over_rtx, false);
   }
 
@@ skipping 234 matching lines @@
   int64_t diff_ms = now_ms - capture_time_ms;
   UpdateTransmissionTimeOffset(buffer_to_send_ptr, length, rtp_header,
                                diff_ms);
   UpdateAbsoluteSendTime(buffer_to_send_ptr, length, rtp_header, now_ms);
 
   PacketOptions options;
   if (AllocateTransportSequenceNumber(&options.packet_id)) {
     if (UpdateTransportSequenceNumber(options.packet_id, buffer_to_send_ptr,
                                       length, rtp_header)) {
       if (transport_feedback_observer_)
-        transport_feedback_observer_->AddPacket(options.packet_id, length,
-                                                true);
+        transport_feedback_observer_->AddPacket(options.packet_id, length);
     }
   }
 
   if (!is_retransmit && !send_over_rtx) {
     UpdateDelayStatistics(capture_time_ms, now_ms);
     UpdateOnSendPacket(options.packet_id, capture_time_ms, rtp_header.ssrc);
   }
 
   bool ret = SendPacketToNetwork(buffer_to_send_ptr, length, options);
   if (ret) {
@@ skipping 108 matching lines @@
                                "capture_time_ms", corrected_time_ms);
     }
     return 0;
   }
 
   PacketOptions options;
   if (AllocateTransportSequenceNumber(&options.packet_id)) {
     if (UpdateTransportSequenceNumber(options.packet_id, buffer, length,
                                       rtp_header)) {
       if (transport_feedback_observer_)
-        transport_feedback_observer_->AddPacket(options.packet_id, length,
-                                                true);
+        transport_feedback_observer_->AddPacket(options.packet_id, length);
     }
   }
   UpdateDelayStatistics(capture_time_ms, now_ms);
   UpdateOnSendPacket(options.packet_id, capture_time_ms, rtp_header.ssrc);
 
   bool sent = SendPacketToNetwork(buffer, length, options);
 
   // Mark the packet as sent in the history even if send failed. Dropping a
   // packet here should be treated as any other packet drop so we should be
   // ready for a retransmission.
@@ skipping 834 matching lines @@
   rtc::CritScope lock(&send_critsect_);
 
   RtpState state;
   state.sequence_number = sequence_number_rtx_;
   state.start_timestamp = start_timestamp_;
 
   return state;
 }
 
 }  // namespace webrtc