[NOT FOR REVIEW] 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 82ef076d41b01d70af7a60f55d4d22d2c2baaf41..f82a1ccf6c37b00188e3abeb0d1ebdc6c7d2bcdc 100644
--- a/webrtc/voice_engine/utility.cc
+++ b/webrtc/voice_engine/utility.cc
@@ -27,7 +27,7 @@ void RemixAndResample(const AudioFrame& src_frame,
                       PushResampler<int16_t>* resampler,
                       AudioFrame* dst_frame) {
   const int16_t* audio_ptr = src_frame.data_;
-  int audio_ptr_num_channels = src_frame.num_channels_;
+  size_t audio_ptr_num_channels = src_frame.num_channels_;
   int16_t mono_audio[AudioFrame::kMaxDataSizeSamples];
 
   // Downmix before resampling.
@@ -47,16 +47,15 @@ void RemixAndResample(const AudioFrame& src_frame,
     assert(false);
   }
 
-  const size_t src_length = src_frame.samples_per_channel_ *
-                         audio_ptr_num_channels;
+  const size_t src_length =
+      src_frame.samples_per_channel_ * audio_ptr_num_channels;
   int out_length = resampler->Resample(audio_ptr, src_length, dst_frame->data_,
                                        AudioFrame::kMaxDataSizeSamples);
   if (out_length == -1) {
     LOG_FERR3(LS_ERROR, Resample, audio_ptr, src_length, 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 (src_frame.num_channels_ == 1 && dst_frame->num_channels_ == 2) {
@@ -73,9 +72,9 @@ void RemixAndResample(const AudioFrame& src_frame,
 
 void DownConvertToCodecFormat(const int16_t* src_data,
                               size_t samples_per_channel,
-                              int num_channels,
+                              size_t num_channels,
                               int sample_rate_hz,
-                              int codec_num_channels,
+                              size_t codec_num_channels,
                               int codec_rate_hz,
                               int16_t* mono_buffer,
                               PushResampler<int16_t>* resampler,
@@ -116,15 +115,15 @@ void DownConvertToCodecFormat(const int16_t* src_data,
     assert(false);
   }
 
-  dst_af->samples_per_channel_ = static_cast<size_t>(out_length / num_channels);
+  dst_af->samples_per_channel_ = out_length / num_channels;
   dst_af->sample_rate_hz_ = destination_rate;
   dst_af->num_channels_ = num_channels;
 }
 
 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);