Match existing type usage better.

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

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 411 matching lines @@
         static_cast<uint32_t>(
             (static_cast<double>(
                  codec_manager_.CurrentEncoder()->SampleRateHz()) /
              static_cast<double>(in_frame.sample_rate_hz_)));
     expected_in_ts_ = in_frame.timestamp_;
   }
 
 
   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;
   }
 
   *ptr_out = &preprocess_frame_;
   preprocess_frame_.num_channels_ = in_frame.num_channels_;
   int16_t audio[WEBRTC_10MS_PCM_AUDIO];
   const int16_t* src_ptr_audio = in_frame.data_;
   int16_t* dest_ptr_audio = preprocess_frame_.data_;
   if (down_mix) {
@@ skipping 24 matching lines @@
 
     if (preprocess_frame_.samples_per_channel_ < 0) {
       WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
                    "Cannot add 10 ms audio, resampling failed");
       return -1;
     }
     preprocess_frame_.sample_rate_hz_ =
         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;
 }
 
 /////////////////////////////////////////
 //   (RED) Redundant Coding
 //
 
 bool AudioCodingModuleImpl::REDStatus() const {
   CriticalSectionScoped lock(acm_crit_sect_);
@@ skipping 765 matching lines @@
       *channels = 1;
       break;
 #endif
     default:
       FATAL() << "Codec type " << codec_type << " not supported.";
   }
   return true;
 }
 
 }  // namespace webrtc