Delete old video defines in engine config and rename file

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.
  */
 
 #ifndef WEBRTC_VIDEO_RTP_STREAM_RECEIVER_H_
 #define WEBRTC_VIDEO_RTP_STREAM_RECEIVER_H_
 
 #include <list>
 #include <memory>
 #include <string>
 #include <vector>
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/criticalsection.h"
-#include "webrtc/engine_configurations.h"
 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
 #include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/video_coding/include/video_coding_defines.h"
 #include "webrtc/typedefs.h"
 #include "webrtc/video_receive_stream.h"
 
 namespace webrtc {
@@ skipping 118 matching lines @@
   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_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_RTP_STREAM_RECEIVER_H_