[NOT FOR REVIEW] Convert channel counts to size_t.

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 95 matching lines @@
 
   void SetTargetBitrate(uint32_t bitrate);
   uint32_t GetTargetBitrate();
 
   // Includes size of RTP and FEC headers.
   size_t MaxDataPayloadLength() const override;
 
   int32_t RegisterPayload(
       const char payload_name[RTP_PAYLOAD_NAME_SIZE],
       const int8_t payload_type, const uint32_t frequency,
-      const uint8_t channels, const uint32_t rate);
+      const size_t channels, const uint32_t rate);
 
   int32_t DeRegisterSendPayload(const int8_t payload_type);
 
   void SetSendPayloadType(int8_t payload_type);
 
   int8_t SendPayloadType() const;
 
   int SendPayloadFrequency() const;
 
   void SetSendingStatus(bool enabled);
@@ skipping 303 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_