Delete unused member RTCPSender::FeedbackState::send_payload_type.

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 258 matching lines @@
 
   if (payload_type_map_.end() == it) {
     return -1;
   }
   RtpUtility::Payload* payload = it->second;
   delete payload;
   payload_type_map_.erase(it);
   return 0;
 }
 
+// TODO(nisse): Delete this method, only used internally and by test code.
 void RTPSender::SetSendPayloadType(int8_t payload_type) {
   rtc::CritScope lock(&send_critsect_);
   payload_type_ = payload_type;
 }
 
-int8_t RTPSender::SendPayloadType() const {
-  rtc::CritScope lock(&send_critsect_);
-  return payload_type_;
-}
-
 void RTPSender::SetMaxRtpPacketSize(size_t max_packet_size) {
   // Sanity check.
   RTC_DCHECK(max_packet_size >= 100 && max_packet_size <= IP_PACKET_SIZE)
       << "Invalid max payload length: " << max_packet_size;
   rtc::CritScope lock(&send_critsect_);
   max_packet_size_ = max_packet_size;
 }
 
 size_t RTPSender::MaxPayloadSize() const {
   if (audio_configured_) {
@@ skipping 975 matching lines @@
     if (rtp_overhead_bytes_per_packet_ == packet.headers_size()) {
       return;
     }
     rtp_overhead_bytes_per_packet_ = packet.headers_size();
     overhead_bytes_per_packet = rtp_overhead_bytes_per_packet_;
   }
   overhead_observer_->OnOverheadChanged(overhead_bytes_per_packet);
 }
 
 }  // namespace webrtc