Unify FrameType and VideoFrameType.

webrtc/modules/rtp_rtcp/source/rtp_sender_audio.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 190 matching lines @@
     }
   }
   if (dtmfToneStarted) {
     if (_audioFeedback)
       _audioFeedback->OnPlayTelephoneEvent(key, dtmfLengthMS, _dtmfLevel);
   }
 
   // A source MAY send events and coded audio packets for the same time
   // but we don't support it
   if (_dtmfEventIsOn) {
-    if (frameType == kFrameEmpty) {
-      // kFrameEmpty is used to drive the DTMF when in CN mode
+    if (frameType == kEmptyFrame) {
+      // kEmptyFrame is used to drive the DTMF when in CN mode
       // it can be triggered more frequently than we want to send the
       // DTMF packets.
       if (packet_size_samples > (captureTimeStamp - _dtmfTimestampLastSent)) {
         // not time to send yet
         return 0;
       }
     }
     _dtmfTimestampLastSent = captureTimeStamp;
     uint32_t dtmfDurationSamples = captureTimeStamp - _dtmfTimestamp;
     bool ended = false;
@@ skipping 29 matching lines @@
                                      !_dtmfEventFirstPacketSent) != 0) {
           return -1;
         }
         _dtmfEventFirstPacketSent = true;
         return 0;
       }
     }
     return 0;
   }
   if (payloadSize == 0 || payloadData == NULL) {
-    if (frameType == kFrameEmpty) {
+    if (frameType == kEmptyFrame) {
       // we don't send empty audio RTP packets
       // no error since we use it to drive DTMF when we use VAD
       return 0;
     }
     return -1;
   }
   uint8_t dataBuffer[IP_PACKET_SIZE];
   bool markerBit = MarkerBit(frameType, payloadType);
 
   int32_t rtpHeaderLength = 0;
@@ skipping 206 matching lines @@
         retVal = _rtpSender->SendToNetwork(dtmfbuffer, 4, 12, -1,
                                            kAllowRetransmission,
                                            RtpPacketSender::kHighPriority);
         sendCount--;
 
     }while (sendCount > 0 && retVal == 0);
 
     return retVal;
 }
 }  // namespace webrtc