Convert channel counts to size_t.

webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h

 /*
  *  Copyright (c) 2014 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 70 matching lines @@
 size_t AudioEncoderIsacT<T>::MaxEncodedBytes() const {
   return kSufficientEncodeBufferSizeBytes;
 }
 
 template <typename T>
 int AudioEncoderIsacT<T>::SampleRateHz() const {
   return T::EncSampRate(isac_state_);
 }
 
 template <typename T>
-int AudioEncoderIsacT<T>::NumChannels() const {
+size_t AudioEncoderIsacT<T>::NumChannels() const {
   return 1;
 }
 
 template <typename T>
 size_t AudioEncoderIsacT<T>::Num10MsFramesInNextPacket() const {
   const int samples_in_next_packet = T::GetNewFrameLen(isac_state_);
   return static_cast<size_t>(
       rtc::CheckedDivExact(samples_in_next_packet,
                            rtc::CheckedDivExact(SampleRateHz(), 100)));
 }
@@ skipping 79 matching lines @@
   // we get an encoding that isn't bit-for-bit identical with what a combined
   // encoder+decoder object produces.
   RTC_CHECK_EQ(0, T::SetDecSampRate(isac_state_, config.sample_rate_hz));
 
   config_ = config;
 }
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_IMPL_H_