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 11 matching lines @@
 class RTPSenderAudio : public DTMFqueue {
  public:
   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,
+                               size_t channels,
                                uint32_t rate,
                                RtpUtility::Payload** payload);
 
   int32_t SendAudio(FrameType frameType,
                     int8_t payloadType,
                     uint32_t captureTimeStamp,
                     const uint8_t* payloadData,
                     size_t payloadSize,
                     const RTPFragmentationHeader* fragmentation);
 
@@ skipping 57 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_