Convert channel counts to size_t.

webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.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 17 matching lines @@
 class TransportFeedback;
 }
 
 const int kVideoPayloadTypeFrequency = 90000;
 
 // Minimum RTP header size in bytes.
 const uint8_t kRtpHeaderSize = 12;
 
 struct AudioPayload {
     uint32_t    frequency;
-    uint8_t     channels;
+    size_t      channels;
     uint32_t    rate;
 };
 
 struct VideoPayload {
     RtpVideoCodecTypes   videoCodecType;
     uint32_t       maxRate;
 };
 
 union PayloadUnion {
     AudioPayload Audio;
@@ skipping 154 matching lines @@
   virtual ~RtpFeedback() {}
 
   // Receiving payload change or SSRC change. (return success!)
   /*
   *   channels    - number of channels in codec (1 = mono, 2 = stereo)
   */
   virtual int32_t OnInitializeDecoder(
       const int8_t payloadType,
       const char payloadName[RTP_PAYLOAD_NAME_SIZE],
       const int frequency,
-      const uint8_t channels,
+      const size_t channels,
       const uint32_t rate) = 0;
 
   virtual void OnIncomingSSRCChanged(const uint32_t ssrc) = 0;
 
   virtual void OnIncomingCSRCChanged(const uint32_t CSRC, const bool added) = 0;
 };
 
 class RtpAudioFeedback {
  public:
   virtual void OnPlayTelephoneEvent(const uint8_t event,
@@ skipping 102 matching lines @@
 };
 
 // Null object version of RtpFeedback.
 class NullRtpFeedback : public RtpFeedback {
  public:
   virtual ~NullRtpFeedback() {}
 
   int32_t OnInitializeDecoder(const int8_t payloadType,
                               const char payloadName[RTP_PAYLOAD_NAME_SIZE],
                               const int frequency,
-                              const uint8_t channels,
+                              const size_t channels,
                               const uint32_t rate) override {
     return 0;
   }
 
   void OnIncomingSSRCChanged(const uint32_t ssrc) override {}
   void OnIncomingCSRCChanged(const uint32_t CSRC, const bool added) override {}
 };
 
 // Null object version of RtpData.
 class NullRtpData : public RtpData {
@@ skipping 60 matching lines @@
 class TransportSequenceNumberAllocator {
  public:
   TransportSequenceNumberAllocator() {}
   virtual ~TransportSequenceNumberAllocator() {}
 
   virtual uint16_t AllocateSequenceNumber() = 0;
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_