Update audio code to use size_t more correctly,

webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.cc

Index: webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.cc
diff --git a/webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.cc b/webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.cc
index 47450bc88703d19890aaefc7ee0b67016cf6648f..d69918b7faf89dcb885a12902b124fb90cd2dd24 100644
--- a/webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.cc
@@ -26,14 +26,11 @@ bool ResampleInputAudioFile::Read(size_t samples,
   if (!InputAudioFile::Read(samples_to_read, temp_destination.get()))
     return false;
   resampler_.ResetIfNeeded(file_rate_hz_, output_rate_hz, 1);
-  int output_length = 0;
-  CHECK_EQ(resampler_.Push(temp_destination.get(),
-                           static_cast<int>(samples_to_read),
-                           destination,
-                           static_cast<int>(samples),
-                           output_length),
+  size_t output_length = 0;
+  CHECK_EQ(resampler_.Push(temp_destination.get(), samples_to_read, destination,
+                           samples, output_length),
            0);
-  CHECK_EQ(static_cast<int>(samples), output_length);
+  CHECK_EQ(samples, output_length);
   return true;
 }