Style cleanups in RtpSender.

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h

 /*
  *  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 94 matching lines @@
 
   bool Sending() const override;
 
   // Drops or relays media packets.
   void SetSendingMediaStatus(bool sending) override;
 
   bool SendingMedia() const override;
 
   // Used by the codec module to deliver a video or audio frame for
   // packetization.
-  int32_t 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 = NULL,
-                           const RTPVideoHeader* rtp_video_hdr = NULL) override;
+  int32_t 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 = NULL,
+      const RTPVideoHeader* rtp_video_header = NULL) override;
 
   bool TimeToSendPacket(uint32_t ssrc,
                         uint16_t sequence_number,
                         int64_t capture_time_ms,
                         bool retransmission,
                         int probe_cluster_id) override;
 
   // Returns the number of padding bytes actually sent, which can be more or
   // less than |bytes|.
   size_t TimeToSendPadding(size_t bytes, int probe_cluster_id) override;
@@ skipping 232 matching lines @@
   PacketLossStats receive_loss_stats_;
 
   // The processed RTT from RtcpRttStats.
   rtc::CriticalSection critical_section_rtt_;
   int64_t rtt_ms_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_