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 43 matching lines @@
   // 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 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;
 
   // Returns the rate at which the RTP timestamps are updated. The default
   // implementation returns SampleRateHz().
   virtual int RtpTimestampRateHz() const;
 
   // Returns the number of 10 ms frames the encoder will put in the next
   // packet. This value may only change when Encode() outputs a packet; i.e.,
   // the encoder may vary the number of 10 ms frames from packet to packet, but
   // it must decide the length of the next packet no later than when outputting
   // the preceding packet.
@@ skipping 59 matching lines @@
   // does nothing.
   virtual void SetProjectedPacketLossRate(double fraction);
 
   // Tells the encoder what average bitrate we'd like it to produce. The
   // encoder is free to adjust or disregard the given bitrate (the default
   // implementation does the latter).
   virtual void SetTargetBitrate(int target_bps);
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_