Match existing type usage better.

webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc

Index: webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
index 0da2eb5c068830c521c168c5d774eb40fe94461a..f417b6ae6da5f93bf22d8d4ec089e08b14e4bf3a 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
@@ -429,8 +429,8 @@ int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame,
 
   if (!down_mix && !resample) {
     // No pre-processing is required.
-    expected_in_ts_ += in_frame.samples_per_channel_;
-    expected_codec_ts_ += in_frame.samples_per_channel_;
+    expected_in_ts_ += static_cast<uint32_t>(in_frame.samples_per_channel_);
+    expected_codec_ts_ += static_cast<uint32_t>(in_frame.samples_per_channel_);
     *ptr_out = &in_frame;
     return 0;
   }
@@ -475,8 +475,9 @@ int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame,
         codec_manager_.CurrentEncoder()->SampleRateHz();
   }
 
-  expected_codec_ts_ += preprocess_frame_.samples_per_channel_;
-  expected_in_ts_ += in_frame.samples_per_channel_;
+  expected_codec_ts_ +=
+      static_cast<uint32_t>(preprocess_frame_.samples_per_channel_);
+  expected_in_ts_ += static_cast<uint32_t>(in_frame.samples_per_channel_);
 
   return 0;
 }