Style cleanups in RtpSender.

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 be8ab34a2771117680367ff7e1e4653ec4c3ab01..b3590a0a7944428e761b99471bb28960ab1fd782 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -46,24 +46,7 @@ RTPExtensionType StringToRtpExtensionType(const std::string& extension) {
 }
 
 RtpRtcp::Configuration::Configuration()
-    : audio(false),
-      receiver_only(false),
-      clock(nullptr),
-      receive_statistics(NullObjectReceiveStatistics()),
-      outgoing_transport(nullptr),
-      intra_frame_callback(nullptr),
-      bandwidth_callback(nullptr),
-      transport_feedback_callback(nullptr),
-      rtt_stats(nullptr),
-      rtcp_packet_type_counter_observer(nullptr),
-      remote_bitrate_estimator(nullptr),
-      paced_sender(nullptr),
-      transport_sequence_number_allocator(nullptr),
-      send_bitrate_observer(nullptr),
-      send_frame_count_observer(nullptr),
-      send_side_delay_observer(nullptr),
-      event_log(nullptr),
-      send_packet_observer(nullptr) {}
+    : receive_statistics(NullObjectReceiveStatistics()) {}
 
 RtpRtcp* RtpRtcp::CreateRtpRtcp(const RtpRtcp::Configuration& configuration) {
   if (configuration.clock) {
@@ -416,7 +399,7 @@ int32_t ModuleRtpRtcpImpl::SendOutgoingData(
     const uint8_t* payload_data,
     size_t payload_size,
     const RTPFragmentationHeader* fragmentation,
-    const RTPVideoHeader* rtp_video_hdr) {
+    const RTPVideoHeader* rtp_video_header) {
   rtcp_sender_.SetLastRtpTime(time_stamp, capture_time_ms);
   // Make sure an RTCP report isn't queued behind a key frame.
   if (rtcp_sender_.TimeToSendRTCPReport(kVideoFrameKey == frame_type)) {
@@ -424,7 +407,7 @@ int32_t ModuleRtpRtcpImpl::SendOutgoingData(
   }
   return rtp_sender_.SendOutgoingData(
       frame_type, payload_type, time_stamp, capture_time_ms, payload_data,
-      payload_size, fragmentation, rtp_video_hdr);
+      payload_size, fragmentation, rtp_video_header);
 }
 
 bool ModuleRtpRtcpImpl::TimeToSendPacket(uint32_t ssrc,