Unify FrameType and VideoFrameType.

webrtc/modules/audio_coding/main/test/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 24 matching lines @@
       _seqNo++ : static_cast<uint16_t>(external_sequence_number_);
   rtpInfo.header.payloadType = payloadType;
   rtpInfo.header.timestamp = (external_send_timestamp_ < 0) ? timeStamp :
       static_cast<uint32_t>(external_send_timestamp_);
 
   if (frameType == kAudioFrameCN) {
     rtpInfo.type.Audio.isCNG = true;
   } else {
     rtpInfo.type.Audio.isCNG = false;
   }
-  if (frameType == kFrameEmpty) {
+  if (frameType == kEmptyFrame) {
     // When frame is empty, we should not transmit it. The frame size of the
     // next non-empty frame will be based on the previous frame size.
     _useLastFrameSize = _lastFrameSizeSample > 0;
     return 0;
   }
 
   rtpInfo.type.Audio.channel = 1;
   // Treat fragmentation separately
   if (fragmentation != NULL) {
     // If silence for too long, send only new data.
@@ skipping 359 matching lines @@
 double Channel::BitRate() {
   double rate;
   uint64_t currTime = TickTime::MillisecondTimestamp();
   _channelCritSect->Enter();
   rate = ((double) _totalBytes * 8.0) / (double) (currTime - _beginTime);
   _channelCritSect->Leave();
   return rate;
 }
 
 }  // namespace webrtc