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

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 0c67c6920f87059c348ee855cc34a6a39a90a7f5..2ddc3564161ca1e1195eeb9f8b02b293871c4e67 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -672,8 +672,8 @@ size_t RTPSender::SendPadData(size_t bytes,
       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;
@@ -934,7 +934,7 @@ bool RTPSender::PrepareAndSendPacket(uint8_t* buffer,
     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,