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

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 07889f9eca5b82ede8e9d7ac5f07aed5a0e42728..9ad9f0464488fb15ec0fa70d7d67502522cd6bbd 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -91,8 +91,6 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
       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),
@@ -101,8 +99,6 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
       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);
@@ -253,7 +249,6 @@ int32_t ModuleRtpRtcpImpl::RegisterSendPayload(
 }
 
 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);
 }
@@ -882,10 +877,6 @@ void ModuleRtpRtcpImpl::BitrateSent(uint32_t* total_rate,
   *nack_rate = rtp_sender_.NackOverheadRate();
 }
 
-void ModuleRtpRtcpImpl::OnRequestIntraFrame() {
-  RequestKeyFrame();
-}
-
 void ModuleRtpRtcpImpl::OnRequestSendReport() {
   SendRTCP(kRtcpSr);
 }