Remove old WebRTC-NewVideoJitterBuffer used for testing the NackModule.

webrtc/modules/video_coding/jitter_buffer.h

 /*
  *  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 184 matching lines @@
   // Returns a list of the sequence numbers currently missing.
   std::vector<uint16_t> GetNackList(bool* request_key_frame);
 
   // Set decode error mode - Should not be changed in the middle of the
   // session. Changes will not influence frames already in the buffer.
   void SetDecodeErrorMode(VCMDecodeErrorMode error_mode);
   VCMDecodeErrorMode decode_error_mode() const { return decode_error_mode_; }
 
   void RegisterStatsCallback(VCMReceiveStatisticsCallback* callback);
 
-  int64_t TimeUntilNextProcess();
-  void Process();
-
  private:
   class SequenceNumberLessThan {
    public:
     bool operator()(const uint16_t& sequence_number1,
                     const uint16_t& sequence_number2) const {
       return IsNewerSequenceNumber(sequence_number2, sequence_number1);
     }
   };
   typedef std::set<uint16_t, SequenceNumberLessThan> SequenceNumberSet;
 
@@ skipping 151 matching lines @@
   int max_packet_age_to_nack_;  // Measured in sequence numbers.
   int max_incomplete_time_ms_;
 
   VCMDecodeErrorMode decode_error_mode_;
   // Estimated rolling average of packets per frame
   float average_packets_per_frame_;
   // average_packets_per_frame converges fast if we have fewer than this many
   // frames.
   int frame_counter_;
 
-  std::unique_ptr<NackModule> nack_module_;
-
   RTC_DISALLOW_COPY_AND_ASSIGN(VCMJitterBuffer);
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_VIDEO_CODING_JITTER_BUFFER_H_