Convert channel counts to size_t.

webrtc/modules/audio_device/android/opensles_player.h

 /*
  *  Copyright (c) 2015 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 76 matching lines @@
   // attached to the Dalvik VM.
   static void SimpleBufferQueueCallback(SLAndroidSimpleBufferQueueItf caller,
                                         void* context);
   void FillBufferQueue();
   // Reads audio data in PCM format using the AudioDeviceBuffer.
   // Can be called both on the main thread (during Start()) and from the
   // internal audio thread while output streaming is active.
   void EnqueuePlayoutData();
 
   // Configures the SL_DATAFORMAT_PCM structure.
-  SLDataFormat_PCM CreatePCMConfiguration(int channels,
+  SLDataFormat_PCM CreatePCMConfiguration(size_t channels,
                                           int sample_rate,
                                           size_t bits_per_sample);
 
   // Allocate memory for audio buffers which will be used to render audio
   // via the SLAndroidSimpleBufferQueueItf interface.
   void AllocateDataBuffers();
 
   // Creates/destroys the main engine object and the SLEngineItf interface.
   bool CreateEngine();
   void DestroyEngine();
@@ skipping 88 matching lines @@
   // properties. This interface is supported on the Audio Player object.
   SLVolumeItf volume_;
 
   // Last time the OpenSL ES layer asked for audio data to play out.
   uint32_t last_play_time_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_PLAYER_H_