Update audio code to use size_t more correctly, webrtc/common_audio/resampler/

webrtc/common_audio/resampler/include/push_resampler.h

 /*
  *  Copyright (c) 2013 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 15 matching lines @@
   PushResampler();
   virtual ~PushResampler();
 
   // Must be called whenever the parameters change. Free to be called at any
   // time as it is a no-op if parameters have not changed since the last call.
   int InitializeIfNeeded(int src_sample_rate_hz, int dst_sample_rate_hz,
                          int num_channels);
 
   // Returns the total number of samples provided in destination (e.g. 32 kHz,
   // 2 channel audio gives 640 samples).
-  int Resample(const T* src, int src_length, T* dst, int dst_capacity);
+  int Resample(const T* src, size_t src_length, T* dst, size_t dst_capacity);
 
  private:
   rtc::scoped_ptr<PushSincResampler> sinc_resampler_;
   rtc::scoped_ptr<PushSincResampler> sinc_resampler_right_;
   int src_sample_rate_hz_;
   int dst_sample_rate_hz_;
   int num_channels_;
   rtc::scoped_ptr<T[]> src_left_;
   rtc::scoped_ptr<T[]> src_right_;
   rtc::scoped_ptr<T[]> dst_left_;
   rtc::scoped_ptr<T[]> dst_right_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_