[cleanup] Remove unused fields/functions from rtcp module.

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.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 73 matching lines @@
                      configuration.intra_frame_callback,
                      configuration.transport_feedback_callback,
                      this),
       clock_(configuration.clock),
       audio_(configuration.audio),
       collision_detected_(false),
       last_process_time_(configuration.clock->TimeInMilliseconds()),
       last_bitrate_process_time_(configuration.clock->TimeInMilliseconds()),
       last_rtt_process_time_(configuration.clock->TimeInMilliseconds()),
       packet_overhead_(28),                     // IPV4 UDP.
-      padding_index_(static_cast<size_t>(-1)),  // Start padding at first child.
-      nack_method_(kNackOff),
       nack_last_time_sent_full_(0),
       nack_last_time_sent_full_prev_(0),
       nack_last_seq_number_sent_(0),
       key_frame_req_method_(kKeyFrameReqPliRtcp),
       remote_bitrate_(configuration.remote_bitrate_estimator),
       rtt_stats_(configuration.rtt_stats),
       critical_section_rtt_(CriticalSectionWrapper::CreateCriticalSection()),
       rtt_ms_(0) {
-  send_video_codec_.codecType = kVideoCodecUnknown;
-
   // Make sure that RTCP objects are aware of our SSRC.
   uint32_t SSRC = rtp_sender_.SSRC();
   rtcp_sender_.SetSSRC(SSRC);
   SetRtcpReceiverSsrcs(SSRC);
 }
 
 // Returns the number of milliseconds until the module want a worker thread
 // to call Process.
 int64_t ModuleRtpRtcpImpl::TimeUntilNextProcess() {
   const int64_t now = clock_->TimeInMilliseconds();
@@ skipping 130 matching lines @@
     const CodecInst& voice_codec) {
   return rtp_sender_.RegisterPayload(
            voice_codec.plname,
            voice_codec.pltype,
            voice_codec.plfreq,
            voice_codec.channels,
            (voice_codec.rate < 0) ? 0 : voice_codec.rate);
 }
 
 int32_t ModuleRtpRtcpImpl::RegisterSendPayload(const VideoCodec& video_codec) {
-  send_video_codec_ = video_codec;
   return rtp_sender_.RegisterPayload(video_codec.plName, video_codec.plType,
                                      90000, 0, 0);
 }
 
 int32_t ModuleRtpRtcpImpl::DeRegisterSendPayload(const int8_t payload_type) {
   return rtp_sender_.DeRegisterSendPayload(payload_type);
 }
 
 int8_t ModuleRtpRtcpImpl::SendPayloadType() const {
   return rtp_sender_.SendPayloadType();
@@ skipping 608 matching lines @@
 void ModuleRtpRtcpImpl::BitrateSent(uint32_t* total_rate,
                                     uint32_t* video_rate,
                                     uint32_t* fec_rate,
                                     uint32_t* nack_rate) const {
   *total_rate = rtp_sender_.BitrateSent();
   *video_rate = rtp_sender_.VideoBitrateSent();
   *fec_rate = rtp_sender_.FecOverheadRate();
   *nack_rate = rtp_sender_.NackOverheadRate();
 }
 
-void ModuleRtpRtcpImpl::OnRequestIntraFrame() {
-  RequestKeyFrame();
-}
-
 void ModuleRtpRtcpImpl::OnRequestSendReport() {
   SendRTCP(kRtcpSr);
 }
 
 int32_t ModuleRtpRtcpImpl::SendRTCPReferencePictureSelection(
     const uint64_t picture_id) {
   return rtcp_sender_.SendRTCP(
       GetFeedbackState(), kRtcpRpsi, 0, 0, false, picture_id);
 }
 
@@ skipping 79 matching lines @@
 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback(
     StreamDataCountersCallback* callback) {
   rtp_sender_.RegisterRtpStatisticsCallback(callback);
 }
 
 StreamDataCountersCallback*
     ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
   return rtp_sender_.GetRtpStatisticsCallback();
 }
 }  // namespace webrtc