Match existing type usage better.

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

 /*
  *  Copyright (c) 2013 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 443 matching lines @@
   audio_frame->vad_activity_ = previous_audio_activity_;
   SetAudioFrameActivityAndType(vad_enabled_, type, audio_frame);
   previous_audio_activity_ = audio_frame->vad_activity_;
   call_stats_.DecodedByNetEq(audio_frame->speech_type_);
 
   // Computes the RTP timestamp of the first sample in |audio_frame| from
   // |GetPlayoutTimestamp|, which is the timestamp of the last sample of
   // |audio_frame|.
   uint32_t playout_timestamp = 0;
   if (GetPlayoutTimestamp(&playout_timestamp)) {
-    audio_frame->timestamp_ =
-        playout_timestamp - audio_frame->samples_per_channel_;
+    audio_frame->timestamp_ = playout_timestamp -
+        static_cast<uint32_t>(audio_frame->samples_per_channel_);
   } else {
     // Remain 0 until we have a valid |playout_timestamp|.
     audio_frame->timestamp_ = 0;
   }
 
   return 0;
 }
 
 int32_t AcmReceiver::AddCodec(int acm_codec_id,
                               uint8_t payload_type,
@@ skipping 360 matching lines @@
 
 void AcmReceiver::GetDecodingCallStatistics(
     AudioDecodingCallStats* stats) const {
   CriticalSectionScoped lock(crit_sect_.get());
   *stats = call_stats_.GetDecodingStatistics();
 }
 
 }  // namespace acm2
 
 }  // namespace webrtc