Make overuse estimator one dimensional.

webrtc/common_types.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 729 matching lines @@
       ret &= (simulcastStream[i] == other.simulcastStream[i]);
     }
     return ret;
   }
 
   bool operator!=(const VideoCodec& other) const {
     return !(*this == other);
   }
 };
 
-// Bandwidth over-use detector options.  These are used to drive
-// experimentation with bandwidth estimation parameters.
-// See modules/remote_bitrate_estimator/overuse_detector.h
-struct OverUseDetectorOptions {
-  OverUseDetectorOptions()
-      : initial_slope(8.0/512.0),
-        initial_offset(0),
-        initial_e(),
-        initial_process_noise(),
-        initial_avg_noise(0.0),
-        initial_var_noise(50) {
-    initial_e[0][0] = 100;
-    initial_e[1][1] = 1e-1;
-    initial_e[0][1] = initial_e[1][0] = 0;
-    initial_process_noise[0] = 1e-13;
-    initial_process_noise[1] = 1e-2;
-  }
-  double initial_slope;
-  double initial_offset;
-  double initial_e[2][2];
-  double initial_process_noise[2];
-  double initial_avg_noise;
-  double initial_var_noise;
-};
-
 // This structure will have the information about when packet is actually
 // received by socket.
 struct PacketTime {
   PacketTime() : timestamp(-1), not_before(-1) {}
   PacketTime(int64_t timestamp, int64_t not_before)
       : timestamp(timestamp), not_before(not_before) {
   }
 
   int64_t timestamp;   // Receive time after socket delivers the data.
   int64_t not_before;  // Earliest possible time the data could have arrived,
@@ skipping 117 matching lines @@
                                    uint32_t ssrc) = 0;
 };
 
 // RTCP mode to use. Compound mode is described by RFC 4585 and reduced-size
 // RTCP mode is described by RFC 5506.
 enum class RtcpMode { kOff, kCompound, kReducedSize };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_COMMON_TYPES_H_