[rtp_rtcp] fixed lint errors in rtp_rtcp module that are not fixed in other CLs

webrtc/modules/rtp_rtcp/source/rtp_sender_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 13 matching lines @@
   RTPSenderAudio(Clock* clock,
                  RTPSender* rtpSender,
                  RtpAudioFeedback* audio_feedback);
   virtual ~RTPSenderAudio();
 
   int32_t RegisterAudioPayload(const char payloadName[RTP_PAYLOAD_NAME_SIZE],
                                int8_t payloadType,
                                uint32_t frequency,
                                uint8_t channels,
                                uint32_t rate,
-                               RtpUtility::Payload*& payload);
+                               RtpUtility::Payload** payload);
 
   int32_t SendAudio(FrameType frameType,
                     int8_t payloadType,
                     uint32_t captureTimeStamp,
                     const uint8_t* payloadData,
                     size_t payloadSize,
                     const RTPFragmentationHeader* fragmentation);
 
   // set audio packet size, used to determine when it's time to send a DTMF
   // packet in silence (CNG)
   int32_t SetAudioPacketSize(uint16_t packetSizeSamples);
 
   // Store the audio level in dBov for
   // header-extension-for-audio-level-indication.
   // Valid range is [0,100]. Actual value is negative.
   int32_t SetAudioLevel(uint8_t level_dBov);
 
   // Send a DTMF tone using RFC 2833 (4733)
   int32_t SendTelephoneEvent(uint8_t key, uint16_t time_ms, uint8_t level);
 
   int AudioFrequency() const;
 
   // Set payload type for Redundant Audio Data RFC 2198
   int32_t SetRED(int8_t payloadType);
 
   // Get payload type for Redundant Audio Data RFC 2198
-  int32_t RED(int8_t& payloadType) const;
+  int32_t RED(int8_t* payloadType) const;
 
  protected:
   int32_t SendTelephoneEventPacket(
       bool ended,
       int8_t dtmf_payload_type,
       uint32_t dtmfTimeStamp,
       uint16_t duration,
       bool markerBit);  // set on first packet in talk burst
 
   bool MarkerBit(const FrameType frameType, const int8_t payloadType);
@@ skipping 28 matching lines @@
   int8_t _cngFBPayloadType GUARDED_BY(_sendAudioCritsect);
   int8_t _lastPayloadType GUARDED_BY(_sendAudioCritsect);
 
   // Audio level indication
   // (https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/)
   uint8_t _audioLevel_dBov GUARDED_BY(_sendAudioCritsect);
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_