Remove old WebRTC-NewVideoJitterBuffer used for testing the NackModule.

webrtc/modules/video_coding/receiver.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 84 matching lines @@
     jitter_buffer_.Start();
   } else {
     jitter_buffer_.Flush();
   }
 }
 
 void VCMReceiver::UpdateRtt(int64_t rtt) {
   jitter_buffer_.UpdateRtt(rtt);
 }
 
-int64_t VCMReceiver::TimeUntilNextProcess() {
-  return jitter_buffer_.TimeUntilNextProcess();
-}
-
-void VCMReceiver::Process() {
-  jitter_buffer_.Process();
-}
-
 int32_t VCMReceiver::InsertPacket(const VCMPacket& packet) {
   // Insert the packet into the jitter buffer. The packet can either be empty or
   // contain media at this point.
   bool retransmitted = false;
   const VCMFrameBufferEnum ret =
       jitter_buffer_.InsertPacket(packet, &retransmitted);
   if (ret == kOldPacket) {
     return VCM_OK;
   } else if (ret == kFlushIndicator) {
     return VCM_FLUSH_INDICATOR;
@@ skipping 171 matching lines @@
   timing_->set_min_playout_delay(desired_delay_ms);
   return 0;
 }
 
 void VCMReceiver::RegisterStatsCallback(
     VCMReceiveStatisticsCallback* callback) {
   jitter_buffer_.RegisterStatsCallback(callback);
 }
 
 }  // namespace webrtc