Convert channel counts to size_t.

webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.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 18 matching lines @@
     // Note that frame size 40 ms produces encodings with two 20 ms frames in
     // them, and frame size 60 ms consists of two 30 ms frames.
   };
 
   explicit AudioEncoderIlbc(const Config& config);
   explicit AudioEncoderIlbc(const CodecInst& codec_inst);
   ~AudioEncoderIlbc() override;
 
   size_t MaxEncodedBytes() const override;
   int SampleRateHz() const override;
-  int NumChannels() const override;
+  size_t NumChannels() const override;
   size_t Num10MsFramesInNextPacket() const override;
   size_t Max10MsFramesInAPacket() const override;
   int GetTargetBitrate() const override;
   EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
                              rtc::ArrayView<const int16_t> audio,
                              size_t max_encoded_bytes,
                              uint8_t* encoded) override;
   void Reset() override;
 
  private:
   size_t RequiredOutputSizeBytes() const;
 
   static const size_t kMaxSamplesPerPacket = 480;
   const Config config_;
   const size_t num_10ms_frames_per_packet_;
   size_t num_10ms_frames_buffered_;
   uint32_t first_timestamp_in_buffer_;
   int16_t input_buffer_[kMaxSamplesPerPacket];
   IlbcEncoderInstance* encoder_;
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderIlbc);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_