[NOT FOR REVIEW] Convert channel counts to size_t.

webrtc/modules/audio_coding/codecs/audio_encoder.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 53 matching lines @@
   // not smaller than the number of bytes actually produced by the encoder.
   EncodedInfo Encode(uint32_t rtp_timestamp,
                      const int16_t* audio,
                      size_t num_samples_per_channel,
                      size_t max_encoded_bytes,
                      uint8_t* encoded);
 
   // Return the input sample rate in Hz and the number of input channels.
   // These are constants set at instantiation time.
   virtual int SampleRateHz() const = 0;
-  virtual int NumChannels() const = 0;
+  virtual size_t NumChannels() const = 0;
 
   // Return the maximum number of bytes that can be produced by the encoder
   // at each Encode() call. The caller can use the return value to determine
   // the size of the buffer that needs to be allocated. This value is allowed
   // to depend on encoder parameters like bitrate, frame size etc., so if
   // any of these change, the caller of Encode() is responsible for checking
   // that the buffer is large enough by calling MaxEncodedBytes() again.
   virtual size_t MaxEncodedBytes() const = 0;
 
   // Returns the rate with which the RTP timestamps are updated. By default,
@@ skipping 58 matching lines @@
   // Sets the maximum rate which the codec may not exceed for any packet.
   virtual void SetMaxRate(int max_rate_bps) = 0;
 
   // Informs the encoder about the maximum sample rate which the decoder will
   // use when decoding the bitstream. The implementation is free to disregard
   // this hint.
   virtual bool SetMaxPlaybackRate(int frequency_hz) = 0;
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_