Style cleanups in RtpSender.

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 28 matching lines @@
     return kRtpExtensionVideoRotation;
   if (extension == RtpExtension::kTransportSequenceNumberUri)
     return kRtpExtensionTransportSequenceNumber;
   if (extension == RtpExtension::kPlayoutDelayUri)
     return kRtpExtensionPlayoutDelay;
   RTC_NOTREACHED() << "Looking up unsupported RTP extension.";
   return kRtpExtensionNone;
 }
 
 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),
-      retransmission_rate_limiter(nullptr) {}
+    : receive_statistics(NullObjectReceiveStatistics()) {}
 
 RtpRtcp* RtpRtcp::CreateRtpRtcp(const RtpRtcp::Configuration& configuration) {
   if (configuration.clock) {
     return new ModuleRtpRtcpImpl(configuration);
   } else {
     // No clock implementation provided, use default clock.
     RtpRtcp::Configuration configuration_copy;
     memcpy(&configuration_copy, &configuration,
            sizeof(RtpRtcp::Configuration));
     configuration_copy.clock = Clock::GetRealTimeClock();
@@ skipping 160 matching lines @@
     const size_t length) {
   // Allow receive of non-compound RTCP packets.
   RTCPUtility::RTCPParserV2 rtcp_parser(rtcp_packet, length, true);
 
   const bool valid_rtcpheader = rtcp_parser.IsValid();
   if (!valid_rtcpheader) {
     LOG(LS_WARNING) << "Incoming invalid RTCP packet";
     return -1;
   }
   RTCPHelp::RTCPPacketInformation rtcp_packet_information;
-  int32_t ret_val = rtcp_receiver_.IncomingRTCPPacket(
-      rtcp_packet_information, &rtcp_parser);
+  int32_t ret_val =
+      rtcp_receiver_.IncomingRTCPPacket(rtcp_packet_information, &rtcp_parser);
   if (ret_val == 0) {
     rtcp_receiver_.TriggerCallbacksFromRTCPPacket(rtcp_packet_information);
   }
   return ret_val;
 }
 
 int32_t ModuleRtpRtcpImpl::RegisterSendPayload(
     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);
+      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) {
   return rtp_sender_.RegisterPayload(video_codec.plName, video_codec.plType,
                                      90000, 0, 0);
 }
 
 void ModuleRtpRtcpImpl::RegisterVideoSendPayload(int payload_type,
                                                  const char* payload_name) {
   RTC_CHECK_EQ(
@@ skipping 132 matching lines @@
 }
 
 int32_t ModuleRtpRtcpImpl::SendOutgoingData(
     FrameType frame_type,
     int8_t payload_type,
     uint32_t time_stamp,
     int64_t capture_time_ms,
     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)) {
       rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpReport);
   }
   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,
                                          uint16_t sequence_number,
                                          int64_t capture_time_ms,
                                          bool retransmission,
                                          int probe_cluster_id) {
   if (SendingMedia() && ssrc == rtp_sender_.SSRC()) {
     return rtp_sender_.TimeToSendPacket(sequence_number, capture_time_ms,
                                         retransmission, probe_cluster_id);
@@ skipping 554 matching lines @@
 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback(
     StreamDataCountersCallback* callback) {
   rtp_sender_.RegisterRtpStatisticsCallback(callback);
 }
 
 StreamDataCountersCallback*
     ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
   return rtp_sender_.GetRtpStatisticsCallback();
 }
 }  // namespace webrtc