[rtp_rtcp] lint whitespace warning removed from most source/ files

webrtc/modules/rtp_rtcp/source/rtp_sender.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 145 matching lines @@
                            const RTPFragmentationHeader* fragmentation,
                            const RTPVideoHeader* rtp_hdr = NULL);
 
   // RTP header extension
   int32_t SetTransmissionTimeOffset(int32_t transmission_time_offset);
   int32_t SetAbsoluteSendTime(uint32_t absolute_send_time);
   void SetVideoRotation(VideoRotation rotation);
   int32_t SetTransportSequenceNumber(uint16_t sequence_number);
 
   int32_t RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id);
-  virtual bool IsRtpHeaderExtensionRegistered(RTPExtensionType type) override;
+  bool IsRtpHeaderExtensionRegistered(RTPExtensionType type) override;
   int32_t DeregisterRtpHeaderExtension(RTPExtensionType type);
 
   size_t RtpHeaderExtensionTotalLength() const;
 
   uint16_t BuildRTPHeaderExtension(uint8_t* data_buffer, bool marker_bit) const;
 
   uint8_t BuildTransmissionTimeOffsetExtension(uint8_t *data_buffer) const;
   uint8_t BuildAudioLevelExtension(uint8_t* data_buffer) const;
   uint8_t BuildAbsoluteSendTimeExtension(uint8_t* data_buffer) const;
   uint8_t BuildVideoRotationExtension(uint8_t* data_buffer) const;
@@ skipping 18 matching lines @@
                                   size_t extension_length_bytes,
                                   size_t* extension_offset) const
       EXCLUSIVE_LOCKS_REQUIRED(send_critsect_.get());
 
   bool UpdateAudioLevel(uint8_t* rtp_packet,
                         size_t rtp_packet_length,
                         const RTPHeader& rtp_header,
                         bool is_voiced,
                         uint8_t dBov) const;
 
-  virtual bool UpdateVideoRotation(uint8_t* rtp_packet,
-                                   size_t rtp_packet_length,
-                                   const RTPHeader& rtp_header,
-                                   VideoRotation rotation) const override;
+  bool UpdateVideoRotation(uint8_t* rtp_packet,
+                           size_t rtp_packet_length,
+                           const RTPHeader& rtp_header,
+                           VideoRotation rotation) const override;
 
   bool TimeToSendPacket(uint16_t sequence_number, int64_t capture_time_ms,
                         bool retransmission);
   size_t TimeToSendPadding(size_t bytes);
 
   // NACK.
   int SelectiveRetransmissions() const;
   int SetSelectiveRetransmissions(uint8_t settings);
   void OnReceivedNACK(const std::list<uint16_t>& nack_sequence_numbers,
                       int64_t avg_rtt);
@@ skipping 242 matching lines @@
   // SetTargetBitrateKbps or GetTargetBitrateKbps. Also remember
   // that by the time the function returns there is no guarantee
   // that the target bitrate is still valid.
   rtc::scoped_ptr<CriticalSectionWrapper> target_bitrate_critsect_;
   uint32_t target_bitrate_ GUARDED_BY(target_bitrate_critsect_);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_