Style cleanups in RtpSender.

- Renamed variables and some function to comply with style guide.
- Removed default argument values.
- Removed some dead code.
- Cleaned up comments formatting in rtp_rtcp.h

R=danilchap@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/2067673004 .

Cr-Commit-Position: refs/heads/master@{#13565}

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,