Update audio code to use size_t more correctly,

webrtc/modules/audio_coding/neteq/post_decode_vad.cc

 /*
  *  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 27 matching lines @@
 
 void PostDecodeVad::Init() {
   running_ = false;
   if (vad_instance_) {
     WebRtcVad_Init(vad_instance_);
     WebRtcVad_set_mode(vad_instance_, kVadMode);
     running_ = true;
   }
 }
 
-void PostDecodeVad::Update(int16_t* signal, int length,
+void PostDecodeVad::Update(int16_t* signal, size_t length,
                            AudioDecoder::SpeechType speech_type,
                            bool sid_frame,
                            int fs_hz) {
   if (!vad_instance_ || !enabled_) {
     return;
   }
 
   if (speech_type == AudioDecoder::kComfortNoise || sid_frame ||
       fs_hz > 16000) {
     // TODO(hlundin): Remove restriction on fs_hz.
     running_ = false;
     active_speech_ = true;
     sid_interval_counter_ = 0;
   } else if (!running_) {
     ++sid_interval_counter_;
   }
 
   if (sid_interval_counter_ >= kVadAutoEnable) {
     Init();
   }
 
   if (length > 0 && running_) {
-    int vad_sample_index = 0;
+    size_t vad_sample_index = 0;
     active_speech_ = false;
     // Loop through frame sizes 30, 20, and 10 ms.
     for (int vad_frame_size_ms = 30; vad_frame_size_ms >= 10;
         vad_frame_size_ms -= 10) {
-      int vad_frame_size_samples = vad_frame_size_ms * fs_hz / 1000;
+      size_t vad_frame_size_samples =
+          static_cast<size_t>(vad_frame_size_ms * fs_hz / 1000);
       while (length - vad_sample_index >= vad_frame_size_samples) {
         int vad_return = WebRtcVad_Process(
             vad_instance_, fs_hz, &signal[vad_sample_index],
             vad_frame_size_samples);
         active_speech_ |= (vad_return == 1);
         vad_sample_index += vad_frame_size_samples;
       }
     }
   }
 }
 
 }  // namespace webrtc