Wire up transport sequence number / send time callbacks to webrtc via libjingle.

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 654 matching lines @@
     PacketOptions options;
     if (using_transport_seq) {
       options.packet_id =
           UpdateTransportSequenceNumber(padding_packet, length, rtp_header);
     }
 
     if (!SendPacketToNetwork(padding_packet, length, options))
       break;
 
     if (using_transport_seq && transport_feedback_observer_) {
-      transport_feedback_observer_->OnPacketSent(PacketInfo(
-          0, now_ms, options.packet_id, length, true));
+      transport_feedback_observer_->OnSentPacket(
+          PacketInfo(0, now_ms, 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) {
@@ skipping 240 matching lines @@
     options.packet_id =
         UpdateTransportSequenceNumber(buffer_to_send_ptr, length, rtp_header);
   }
 
   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_->OnPacketSent(
+    transport_feedback_observer_->OnSentPacket(
         PacketInfo(0, now_ms, 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,
@@ skipping 943 matching lines @@
   CriticalSectionScoped lock(send_critsect_.get());
 
   RtpState state;
   state.sequence_number = sequence_number_rtx_;
   state.start_timestamp = start_timestamp_;
 
   return state;
 }
 
 }  // namespace webrtc