[NOT FOR REVIEW] Convert channel counts to size_t.

webrtc/modules/audio_coding/codecs/audio_encoder_mutable_impl.h

 /*
  *  Copyright (c) 2015 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 44 matching lines @@
                                                   size_t max_encoded_bytes,
                                                   uint8_t* encoded) override {
     CriticalSectionScoped cs(encoder_lock_.get());
     return encoder_->EncodeInternal(rtp_timestamp, audio, max_encoded_bytes,
                                     encoded);
   }
   int SampleRateHz() const override {
     CriticalSectionScoped cs(encoder_lock_.get());
     return encoder_->SampleRateHz();
   }
-  int NumChannels() const override {
+  size_t NumChannels() const override {
     CriticalSectionScoped cs(encoder_lock_.get());
     return encoder_->NumChannels();
   }
   size_t MaxEncodedBytes() const override {
     CriticalSectionScoped cs(encoder_lock_.get());
     return encoder_->MaxEncodedBytes();
   }
   int RtpTimestampRateHz() const override {
     CriticalSectionScoped cs(encoder_lock_.get());
     return encoder_->RtpTimestampRateHz();
@@ skipping 48 matching lines @@
   const rtc::scoped_ptr<CriticalSectionWrapper> encoder_lock_;
 
  private:
   rtc::scoped_ptr<T> encoder_ GUARDED_BY(encoder_lock_);
   typename T::Config config_ GUARDED_BY(encoder_lock_);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_MUTABLE_IMPL_H_