Convert some more things 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 78 matching lines @@
                                         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,
                                           int sample_rate,
-                                          int bits_per_sample);
+                                          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();
 
   // Creates/destroys the output mix object.
@@ skipping 28 matching lines @@
   bool playing_;
 
   // PCM-type format definition.
   // TODO(henrika): add support for SLAndroidDataFormat_PCM_EX (android-21) if
   // 32-bit float representation is needed.
   SLDataFormat_PCM pcm_format_;
 
   // Number of bytes per audio buffer in each |audio_buffers_[i]|.
   // Typical sizes are 480 or 512 bytes corresponding to native output buffer
   // sizes of 240 or 256 audio frames respectively.
-  int bytes_per_buffer_;
+  size_t bytes_per_buffer_;
 
   // Queue of audio buffers to be used by the player object for rendering
   // audio. They will be used in a Round-robin way and the size of each buffer
   // is given by FineAudioBuffer::RequiredBufferSizeBytes().
   rtc::scoped_ptr<SLint8[]> audio_buffers_[kNumOfOpenSLESBuffers];
 
   // FineAudioBuffer takes an AudioDeviceBuffer which delivers audio data
   // in chunks of 10ms. It then allows for this data to be pulled in
   // a finer or coarser granularity. I.e. interacting with this class instead
   // of directly with the AudioDeviceBuffer one can ask for any number of
@@ skipping 34 matching lines @@
   SLAndroidSimpleBufferQueueItf simple_buffer_queue_;
 
   // This interface exposes controls for manipulating the object’s audio volume
   // properties. This interface is supported on the Audio Player object.
   SLVolumeItf volume_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_PLAYER_H_