Match existing type usage better.

webrtc/voice_engine/channel.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 1079 matching lines @@
     _audioDeviceModulePtr = &audioDeviceModule;
     _voiceEngineObserverPtr = voiceEngineObserver;
     _callbackCritSectPtr = callbackCritSect;
     return 0;
 }
 
 int32_t
 Channel::UpdateLocalTimeStamp()
 {
 
-    _timeStamp += _audioFrame.samples_per_channel_;
+    _timeStamp += static_cast<uint32_t>(_audioFrame.samples_per_channel_);
     return 0;
 }
 
 int32_t
 Channel::StartPlayout()
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::StartPlayout()");
     if (channel_state_.Get().playing)
     {
@@ skipping 2336 matching lines @@
     // This call will trigger AudioPacketizationCallback::SendData if encoding
     // is done and payload is ready for packetization and transmission.
     // Otherwise, it will return without invoking the callback.
     if (audio_coding_->Add10MsData((AudioFrame&)_audioFrame) < 0)
     {
         WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,_channelId),
                      "Channel::EncodeAndSend() ACM encoding failed");
         return 0xFFFFFFFF;
     }
 
-    _timeStamp += _audioFrame.samples_per_channel_;
+    _timeStamp += static_cast<uint32_t>(_audioFrame.samples_per_channel_);
     return 0;
 }
 
 void Channel::DisassociateSendChannel(int channel_id) {
   CriticalSectionScoped lock(assoc_send_channel_lock_.get());
   Channel* channel = associate_send_channel_.channel();
   if (channel && channel->ChannelId() == channel_id) {
     // If this channel is associated with a send channel of the specified
     // Channel ID, disassociate with it.
     ChannelOwner ref(NULL);
@@ skipping 678 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt)
       != 0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc