Revert of Remove SendPacer from ViEEncoder

webrtc/modules/bitrate_controller/include/bitrate_controller.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.
  *
  *  Usage: this class will register multiple RtcpBitrateObserver's one at each
  *  RTCP module. It will aggregate the results and run one bandwidth estimation
  *  and push the result to the encoders via BitrateObserver(s).
  */
 
 #ifndef WEBRTC_MODULES_BITRATE_CONTROLLER_INCLUDE_BITRATE_CONTROLLER_H_
 #define WEBRTC_MODULES_BITRATE_CONTROLLER_INCLUDE_BITRATE_CONTROLLER_H_
 
 #include <map>
 
 #include "webrtc/modules/include/module.h"
-#include "webrtc/modules/pacing/paced_sender.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 
 namespace webrtc {
 
 class CriticalSectionWrapper;
 class RtcEventLog;
 struct PacketInfo;
 
-// Deprecated
-// TODO(perkj): Remove BitrateObserver when no implementations use it.
 class BitrateObserver {
   // Observer class for bitrate changes announced due to change in bandwidth
   // estimate or due to bitrate allocation changes. Fraction loss and rtt is
   // also part of this callback to allow the obsevrer to optimize its settings
   // for different types of network environments. The bitrate does not include
   // packet headers and is measured in bits per second.
  public:
   virtual void OnNetworkChanged(uint32_t bitrate_bps,
                                 uint8_t fraction_loss,  // 0 - 255.
                                 int64_t rtt_ms) = 0;
 
   virtual ~BitrateObserver() {}
 };
 
 class BitrateController : public Module {
   // This class collects feedback from all streams sent to a peer (via
   // RTCPBandwidthObservers). It does one  aggregated send side bandwidth
   // estimation and divide the available bitrate between all its registered
   // BitrateObservers.
  public:
-  static const int kDefaultStartBitratebps = 300000;
+  static const int kDefaultStartBitrateKbps = 300;
 
-  // Deprecated:
-  // TODO(perkj): BitrateObserver has been deprecated and is not used in WebRTC.
-  // Remove this method once other other projects does not use it.
   static BitrateController* CreateBitrateController(Clock* clock,
                                                     BitrateObserver* observer);
-  static BitrateController* CreateBitrateController(Clock* clock);
-
   virtual ~BitrateController() {}
 
   virtual RtcpBandwidthObserver* CreateRtcpBandwidthObserver() = 0;
 
   // Deprecated
   virtual void SetStartBitrate(int start_bitrate_bps) = 0;
   // Deprecated
   virtual void SetMinMaxBitrate(int min_bitrate_bps, int max_bitrate_bps) = 0;
   virtual void SetBitrates(int start_bitrate_bps,
                            int min_bitrate_bps,
                            int max_bitrate_bps) = 0;
 
   virtual void UpdateDelayBasedEstimate(uint32_t bitrate_bps) = 0;
 
   virtual void SetEventLog(RtcEventLog* event_log) = 0;
 
   // Gets the available payload bandwidth in bits per second. Note that
   // this bandwidth excludes packet headers.
   virtual bool AvailableBandwidth(uint32_t* bandwidth) const = 0;
 
   virtual void SetReservedBitrate(uint32_t reserved_bitrate_bps) = 0;
-
-  virtual bool GetNetworkParameters(uint32_t* bitrate,
-                                    uint8_t* fraction_loss,
-                                    int64_t* rtt) = 0;
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_BITRATE_CONTROLLER_INCLUDE_BITRATE_CONTROLLER_H_