Convert channel counts to size_t.

webrtc/voice_engine/utility.cc

Index: webrtc/voice_engine/utility.cc
diff --git a/webrtc/voice_engine/utility.cc b/webrtc/voice_engine/utility.cc
index eb442ecb8f74c8ac607b1f74c85274a0566520b5..605e55369e3d4a31666e17bf69d3b04ccd054630 100644
--- a/webrtc/voice_engine/utility.cc
+++ b/webrtc/voice_engine/utility.cc
@@ -34,12 +34,12 @@ void RemixAndResample(const AudioFrame& src_frame,
 
 void RemixAndResample(const int16_t* src_data,
                       size_t samples_per_channel,
-                      int num_channels,
+                      size_t num_channels,
                       int sample_rate_hz,
                       PushResampler<int16_t>* resampler,
                       AudioFrame* dst_frame) {
   const int16_t* audio_ptr = src_data;
-  int audio_ptr_num_channels = num_channels;
+  size_t audio_ptr_num_channels = num_channels;
   int16_t mono_audio[AudioFrame::kMaxDataSizeSamples];
 
   // Downmix before resampling.
@@ -68,8 +68,7 @@ void RemixAndResample(const int16_t* src_data,
                   << ", dst_frame->data_ = " << dst_frame->data_;
     assert(false);
   }
-  dst_frame->samples_per_channel_ =
-      static_cast<size_t>(out_length / audio_ptr_num_channels);
+  dst_frame->samples_per_channel_ = out_length / audio_ptr_num_channels;
 
   // Upmix after resampling.
   if (num_channels == 1 && dst_frame->num_channels_ == 2) {
@@ -81,9 +80,9 @@ void RemixAndResample(const int16_t* src_data,
 }
 
 void MixWithSat(int16_t target[],
-                int target_channel,
+                size_t target_channel,
                 const int16_t source[],
-                int source_channel,
+                size_t source_channel,
                 size_t source_len) {
   assert(target_channel == 1 || target_channel == 2);
   assert(source_channel == 1 || source_channel == 2);