Convert channel counts to size_t.

webrtc/modules/audio_coding/neteq/include/neteq.h

 /*
  *  Copyright (c) 2012 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 153 matching lines @@
 
   // Instructs NetEq to deliver 10 ms of audio data. The data is written to
   // |output_audio|, which can hold (at least) |max_length| elements.
   // The number of channels that were written to the output is provided in
   // the output variable |num_channels|, and each channel contains
   // |samples_per_channel| elements. If more than one channel is written,
   // the samples are interleaved.
   // The speech type is written to |type|, if |type| is not NULL.
   // Returns kOK on success, or kFail in case of an error.
   virtual int GetAudio(size_t max_length, int16_t* output_audio,
-                       size_t* samples_per_channel, int* num_channels,
+                       size_t* samples_per_channel, size_t* num_channels,
                        NetEqOutputType* type) = 0;
 
   // Associates |rtp_payload_type| with |codec| and |codec_name|, and stores the
   // information in the codec database. Returns 0 on success, -1 on failure.
   // The name is only used to provide information back to the caller about the
   // decoders. Hence, the name is arbitrary, and may be empty.
   virtual int RegisterPayloadType(NetEqDecoder codec,
                                   const std::string& codec_name,
                                   uint8_t rtp_payload_type) = 0;
 
@@ skipping 114 matching lines @@
 
  protected:
   NetEq() {}
 
  private:
   RTC_DISALLOW_COPY_AND_ASSIGN(NetEq);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_INCLUDE_NETEQ_H_