[NOT FOR REVIEW] Convert channel counts to size_t.

webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.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 226 matching lines @@
 
   std::vector<uint16_t> GetNackList(int64_t round_trip_time_ms) const override;
 
   void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const override;
 
  private:
   struct InputData {
     uint32_t input_timestamp;
     const int16_t* audio;
     size_t length_per_channel;
-    uint8_t audio_channel;
+    size_t audio_channel;
     // If a re-mix is required (up or down), this buffer will store a re-mixed
     // version of the input.
     int16_t buffer[WEBRTC_10MS_PCM_AUDIO];
   };
 
   // This member class writes values to the named UMA histogram, but only if
   // the value has changed since the last time (and always for the first call).
   class ChangeLogger {
    public:
     explicit ChangeLogger(const std::string& histogram_name)
@@ skipping 130 matching lines @@
       AudioDecodingCallStats* call_stats) const override;
 
  private:
   // Temporary method to be used during redesign phase.
   // Maps |codec_type| (a value from the anonymous enum in acm2::ACMCodecDB) to
   // |codec_name|, |sample_rate_hz|, and |channels|.
   // TODO(henrik.lundin) Remove this when no longer needed.
   static bool MapCodecTypeToParameters(int codec_type,
                                        std::string* codec_name,
                                        int* sample_rate_hz,
-                                       int* channels);
+                                       size_t* channels);
 
   int playout_frequency_hz_;
   // TODO(henrik.lundin): All members below this line are temporary and should
   // be removed after refactoring is completed.
   rtc::scoped_ptr<acm2::AudioCodingModuleImpl> acm_old_;
   CodecInst current_send_codec_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_AUDIO_CODING_MODULE_IMPL_H_