[NOT FOR REVIEW] Convert channel counts to size_t.

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 @@
 public:
     RTPSenderAudio(const int32_t id,
                    Clock* clock,
                    RTPSender* rtpSender,
                    RtpAudioFeedback* audio_feedback);
     virtual ~RTPSenderAudio();
 
     int32_t RegisterAudioPayload(const char payloadName[RTP_PAYLOAD_NAME_SIZE],
                                  const int8_t payloadType,
                                  const uint32_t frequency,
-                                 const uint8_t channels,
+                                 const size_t channels,
                                  const uint32_t rate,
                                  RtpUtility::Payload*& payload);
 
     int32_t SendAudio(const FrameType frameType,
                       const int8_t payloadType,
                       const uint32_t captureTimeStamp,
                       const uint8_t* payloadData,
                       const size_t payloadSize,
                       const RTPFragmentationHeader* fragmentation);
 
@@ skipping 58 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_