[NOT FOR REVIEW] Convert channel counts to size_t.

webrtc/modules/rtp_rtcp/interface/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 15 matching lines @@
 namespace webrtc {
 
 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
@@ skipping 189 matching lines @@
 
     // Receiving payload change or SSRC change. (return success!)
     /*
     *   channels    - number of channels in codec (1 = mono, 2 = stereo)
     */
     virtual int32_t OnInitializeDecoder(
         const int32_t id,
         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 int32_t id,
                                         const uint32_t ssrc) = 0;
 
     virtual void OnIncomingCSRCChanged( const int32_t id,
                                         const uint32_t CSRC,
                                         const bool added) = 0;
 };
 
@@ skipping 47 matching lines @@
 
 // Null object version of RtpFeedback.
 class NullRtpFeedback : public RtpFeedback {
  public:
   virtual ~NullRtpFeedback() {}
 
   int32_t OnInitializeDecoder(const int32_t id,
                               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 int32_t id, const uint32_t ssrc) override {}
 
   void OnIncomingCSRCChanged(const int32_t id,
                              const uint32_t CSRC,
                              const bool added) override {}
 };
@@ skipping 33 matching lines @@
   uint64_t single_packet_loss_count;
   // The number of events in which more than one adjacent packet was lost.
   uint64_t multiple_packet_loss_event_count;
   // The number of packets lost in events where more than one adjacent packet
   // was lost.
   uint64_t multiple_packet_loss_packet_count;
 };
 
 }  // namespace webrtc
 #endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_