[NOT FOR REVIEW] Convert channel counts to size_t.

talk/media/webrtc/webrtcvoiceengine.cc

 /*
  * libjingle
  * Copyright 2004 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 46 matching lines @@
 #ifdef WIN32
 #include <objbase.h>  // NOLINT
 #endif
 
 namespace cricket {
 
 static const int kMaxNumPacketSize = 6;
 struct CodecPref {
   const char* name;
   int clockrate;
-  int channels;
+  size_t channels;
   int payload_type;
   bool is_multi_rate;
   int packet_sizes_ms[kMaxNumPacketSize];
 };
 // Note: keep the supported packet sizes in ascending order.
 static const CodecPref kCodecPrefs[] = {
   { kOpusCodecName,   48000, 2, 111, true,  { 10, 20, 40, 60 } },
   { kIsacCodecName,   16000, 1, 103, true,  { 30, 60 } },
   { kIsacCodecName,   32000, 1, 104, true,  { 30 } },
   // G722 should be advertised as 8000 Hz because of the RFC "bug".
@@ skipping 1579 matching lines @@
     renderer_->RemoveChannel(channel_);
     renderer_->SetSink(NULL);
     renderer_ = NULL;
   }
 
   // AudioRenderer::Sink implementation.
   // This method is called on the audio thread.
   void OnData(const void* audio_data,
               int bits_per_sample,
               int sample_rate,
-              int number_of_channels,
+              size_t number_of_channels,
               size_t number_of_frames) override {
     voe_audio_transport_->OnData(channel_,
                                  audio_data,
                                  bits_per_sample,
                                  sample_rate,
                                  number_of_channels,
                                  number_of_frames);
   }
 
   // Callback from the |renderer_| when it is going away. In case Start() has
@@ skipping 1989 matching lines @@
 
 int WebRtcSoundclipStream::Rewind() {
   mem_.Rewind();
   // Return -1 to keep VoiceEngine from looping.
   return (loop_) ? 0 : -1;
 }
 
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE