Update audio code to use size_t more correctly, webrtc/voice_engine/ portion.

webrtc/voice_engine/utility.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 24 matching lines @@
 // Downmix and downsample the audio in |src_data| to |dst_af| as necessary,
 // specified by |codec_num_channels| and |codec_rate_hz|. |mono_buffer| is
 // temporary space and must be of sufficient size to hold the downmixed source
 // audio (recommend using a size of kMaxMonoDataSizeSamples).
 //
 // |dst_af| will have its data and format members (sample rate, channels and
 // samples per channel) set appropriately. No other members will be changed.
 // TODO(ajm): For now, this still calls Reset() on |dst_af|. Remove this, as
 // it shouldn't be needed.
 void DownConvertToCodecFormat(const int16_t* src_data,
-                              int samples_per_channel,
+                              size_t samples_per_channel,
                               int num_channels,
                               int sample_rate_hz,
                               int codec_num_channels,
                               int codec_rate_hz,
                               int16_t* mono_buffer,
                               PushResampler<int16_t>* resampler,
                               AudioFrame* dst_af);
 
 void MixWithSat(int16_t target[],
                 int target_channel,
                 const int16_t source[],
                 int source_channel,
-                int source_len);
+                size_t source_len);
 
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_UTILITY_H_