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

webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.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 24 matching lines @@
   // The following three methods implement the TelephoneEventHandler interface.
   // Forward DTMFs to decoder for playout.
   void SetTelephoneEventForwardToDecoder(bool forward_to_decoder);
 
   // Is forwarding of outband telephone events turned on/off?
   bool TelephoneEventForwardToDecoder() const;
 
   // Is TelephoneEvent configured with payload type payload_type
   bool TelephoneEventPayloadType(const int8_t payload_type) const;
 
-  TelephoneEventHandler* GetTelephoneEventHandler() {
-    return this;
-  }
+  TelephoneEventHandler* GetTelephoneEventHandler() { return this; }
 
   // Returns true if CNG is configured with payload type payload_type. If so,
   // the frequency and cng_payload_type_has_changed are filled in.
   bool CNGPayloadType(const int8_t payload_type,
                       uint32_t* frequency,
                       bool* cng_payload_type_has_changed);
 
   int32_t ParseRtpPacket(WebRtcRTPHeader* rtp_header,
                          const PayloadUnion& specific_payload,
                          bool is_red,
@@ skipping 31 matching lines @@
 
   // We need to look out for special payload types here and sometimes reset
   // statistics. In addition we sometimes need to tweak the frequency.
   void CheckPayloadChanged(int8_t payload_type,
                            PayloadUnion* specific_payload,
                            bool* should_discard_changes) override;
 
   int Energy(uint8_t array_of_energy[kRtpCsrcSize]) const override;
 
  private:
-
-  int32_t ParseAudioCodecSpecific(
-      WebRtcRTPHeader* rtp_header,
-      const uint8_t* payload_data,
-      size_t payload_length,
-      const AudioPayload& audio_specific,
-      bool is_red);
+  int32_t ParseAudioCodecSpecific(WebRtcRTPHeader* rtp_header,
+                                  const uint8_t* payload_data,
+                                  size_t payload_length,
+                                  const AudioPayload& audio_specific,
+                                  bool is_red);
 
   uint32_t last_received_frequency_;
 
   bool telephone_event_forward_to_decoder_;
   int8_t telephone_event_payload_type_;
   std::set<uint8_t> telephone_event_reported_;
 
   int8_t cng_nb_payload_type_;
   int8_t cng_wb_payload_type_;
   int8_t cng_swb_payload_type_;
   int8_t cng_fb_payload_type_;
   int8_t cng_payload_type_;
 
   // G722 is special since it use the wrong number of RTP samples in timestamp
   // VS. number of samples in the frame
   int8_t g722_payload_type_;
   bool last_received_g722_;
 
   uint8_t num_energy_;
   uint8_t current_remote_energy_[kRtpCsrcSize];
 
   RtpAudioFeedback* cb_audio_feedback_;
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_