Style cleanups in RtpSender.

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h

Index: webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index ff3f01a21dab7d2299db5be86abadb0c5f77b0f4..e8fc545ec52e415784a7c3fedb16ecca5679de92 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -112,14 +112,15 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
 
   // 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,