Unify FrameType and VideoFrameType.

webrtc/modules/video_coding/main/source/jitter_buffer.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.
  */
 #include "webrtc/modules/video_coding/main/source/jitter_buffer.h"
@@ skipping 625 matching lines @@
   // We are keeping track of the first and latest seq numbers, and
   // the number of wraps to be able to calculate how many packets we expect.
   if (first_packet_since_reset_) {
     // Now it's time to start estimating jitter
     // reset the delay estimate.
     inter_frame_delay_.Reset(now_ms);
   }
 
   // Empty packets may bias the jitter estimate (lacking size component),
   // therefore don't let empty packet trigger the following updates:
-  if (packet.frameType != kFrameEmpty) {
+  if (packet.frameType != kEmptyFrame) {
     if (waiting_for_completion_.timestamp == packet.timestamp) {
       // This can get bad if we have a lot of duplicate packets,
       // we will then count some packet multiple times.
       waiting_for_completion_.frame_size += packet.sizeBytes;
       waiting_for_completion_.latest_packet_time = now_ms;
     } else if (waiting_for_completion_.latest_packet_time >= 0 &&
                waiting_for_completion_.latest_packet_time + 2000 <= now_ms) {
       // A packet should never be more than two seconds late
       UpdateJitterEstimate(waiting_for_completion_, true);
       waiting_for_completion_.latest_packet_time = -1;
@@ skipping 596 matching lines @@
   }
   // Evaluate if the RTT is higher than |high_rtt_nack_threshold_ms_|, and in
   // that case we don't wait for retransmissions.
   if (high_rtt_nack_threshold_ms_ >= 0 &&
       rtt_ms_ >= high_rtt_nack_threshold_ms_) {
     return false;
   }
   return true;
 }
 }  // namespace webrtc