Make the new jitter buffer the default jitter buffer.

webrtc/video/rtp_stream_receiver.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 171 matching lines @@
 
   rtc::CriticalSection receive_cs_;
   bool receiving_ GUARDED_BY(receive_cs_);
   uint8_t restored_packet_[IP_PACKET_SIZE] GUARDED_BY(receive_cs_);
   bool restored_packet_in_use_ GUARDED_BY(receive_cs_);
   int64_t last_packet_log_ms_ GUARDED_BY(receive_cs_);
 
   const std::unique_ptr<RtpRtcp> rtp_rtcp_;
 
   // Members for the new jitter buffer experiment.
-  bool jitter_buffer_experiment_;
   video_coding::OnCompleteFrameCallback* complete_frame_callback_;
   KeyFrameRequestSender* keyframe_request_sender_;
   VCMTiming* timing_;
   std::unique_ptr<NackModule> nack_module_;
   rtc::scoped_refptr<video_coding::PacketBuffer> packet_buffer_;
   std::unique_ptr<video_coding::RtpFrameReferenceFinder> reference_finder_;
   rtc::CriticalSection last_seq_num_cs_;
   std::map<uint16_t, uint16_t, DescendingSeqNumComp<uint16_t>>
       last_seq_num_for_pic_id_ GUARDED_BY(last_seq_num_cs_);
   video_coding::H264SpsPpsTracker tracker_;
   // TODO(johan): Remove pt_codec_params_ once
   // https://bugs.chromium.org/p/webrtc/issues/detail?id=6883 is resolved.
   // Maps a payload type to a map of out-of-band supplied codec parameters.
   std::map<uint8_t, std::map<std::string, std::string>> pt_codec_params_;
   int16_t last_payload_type_ = -1;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_RTP_STREAM_RECEIVER_H_