[NOT FOR REVIEW] Convert channel counts to size_t.

webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.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 25 matching lines @@
  *                                 Favor faithfulness to the original input.
  *
  * Output:
  *      - inst               : a pointer to Encoder context that is created
  *                             if success.
  *
  * Return value              : 0 - Success
  *                            -1 - Error
  */
 int16_t WebRtcOpus_EncoderCreate(OpusEncInst** inst,
-                                 int32_t channels,
+                                 size_t channels,
                                  int32_t application);
 
 int16_t WebRtcOpus_EncoderFree(OpusEncInst* inst);
 
 /****************************************************************************
  * WebRtcOpus_Encode(...)
  *
  * This function encodes audio as a series of Opus frames and inserts
  * it into a packet. Input buffer can be any length.
  *
@@ skipping 272 matching lines @@
  *                                 1 - the packet contains FEC.
  */
 int WebRtcOpus_PacketHasFec(const uint8_t* payload,
                             size_t payload_length_bytes);
 
 #ifdef __cplusplus
 }  // extern "C"
 #endif
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INTERFACE_OPUS_INTERFACE_H_