Update audio code to use size_t more correctly, webrtc/modules/audio_processing/

webrtc/modules/audio_processing/voice_detection_impl.cc

 /*
  *  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 122 matching lines @@
   return frame_size_ms_;
 }
 
 int VoiceDetectionImpl::Initialize() {
   int err = ProcessingComponent::Initialize();
   if (err != apm_->kNoError || !is_component_enabled()) {
     return err;
   }
 
   using_external_vad_ = false;
-  frame_size_samples_ = frame_size_ms_ *
-      apm_->proc_split_sample_rate_hz() / 1000;
+  frame_size_samples_ = static_cast<size_t>(
+      frame_size_ms_ * apm_->proc_split_sample_rate_hz() / 1000);
   // TODO(ajm): intialize frame buffer here.
 
   return apm_->kNoError;
 }
 
 void* VoiceDetectionImpl::CreateHandle() const {
   return WebRtcVad_Create();
 }
 
 void VoiceDetectionImpl::DestroyHandle(void* handle) const {
@@ skipping 12 matching lines @@
 int VoiceDetectionImpl::num_handles_required() const {
   return 1;
 }
 
 int VoiceDetectionImpl::GetHandleError(void* handle) const {
   // The VAD has no get_error() function.
   assert(handle != NULL);
   return apm_->kUnspecifiedError;
 }
 }  // namespace webrtc