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

webrtc/modules/utility/source/file_recorder_impl.cc

Index: webrtc/modules/utility/source/file_recorder_impl.cc
diff --git a/webrtc/modules/utility/source/file_recorder_impl.cc b/webrtc/modules/utility/source/file_recorder_impl.cc
index e86afc69722ab7f06d45e4c90e9e231cebd89348..6a5f2c27c25113bad14135e9af4b523b98c6cdb8 100644
--- a/webrtc/modules/utility/source/file_recorder_impl.cc
+++ b/webrtc/modules/utility/source/file_recorder_impl.cc
@@ -156,7 +156,7 @@ int32_t FileRecorderImpl::RecordAudioToFile(
         tempAudioFrame.sample_rate_hz_ = incomingAudioFrame.sample_rate_hz_;
         tempAudioFrame.samples_per_channel_ =
           incomingAudioFrame.samples_per_channel_;
-        for (uint16_t i = 0;
+        for (size_t i = 0;
              i < (incomingAudioFrame.samples_per_channel_); i++)
         {
             // Sample value is the average of left and right buffer rounded to
@@ -174,7 +174,7 @@ int32_t FileRecorderImpl::RecordAudioToFile(
         tempAudioFrame.sample_rate_hz_ = incomingAudioFrame.sample_rate_hz_;
         tempAudioFrame.samples_per_channel_ =
           incomingAudioFrame.samples_per_channel_;
-        for (uint16_t i = 0;
+        for (size_t i = 0;
              i < (incomingAudioFrame.samples_per_channel_); i++)
         {
             // Duplicate sample to both channels
@@ -210,7 +210,7 @@ int32_t FileRecorderImpl::RecordAudioToFile(
             return -1;
         }
     } else {
-        int outLen = 0;
+        size_t outLen = 0;
         _audioResampler.ResetIfNeeded(ptrAudioFrame->sample_rate_hz_,
                                       codec_info_.plfreq,
                                       ptrAudioFrame->num_channels_);