Remove rtcp_utility as mostly unused.

webrtc/modules/rtp_rtcp/source/rtcp_sender.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 10 matching lines @@
 #include "webrtc/api/call/transport.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/optional.h"
 #include "webrtc/base/random.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_nack_stats.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/dlrr.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmb_item.h"
-#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 class ModuleRtpRtcpImpl;
 class RtcEventLog;
 
 class NACKStringBuilder {
  public:
   NACKStringBuilder();
@@ skipping 206 matching lines @@
       GUARDED_BY(critical_section_rtcp_sender_);
 
   // XR VoIP metric
   rtc::Optional<RTCPVoIPMetric> xr_voip_metric_
       GUARDED_BY(critical_section_rtcp_sender_);
 
   RtcpPacketTypeCounterObserver* const packet_type_counter_observer_;
   RtcpPacketTypeCounter packet_type_counter_
       GUARDED_BY(critical_section_rtcp_sender_);
 
-  RTCPUtility::NackStats nack_stats_ GUARDED_BY(critical_section_rtcp_sender_);
+  RtcpNackStats nack_stats_ GUARDED_BY(critical_section_rtcp_sender_);
 
   rtc::Optional<BitrateAllocation> video_bitrate_allocation_
       GUARDED_BY(critical_section_rtcp_sender_);
 
   void SetFlag(uint32_t type, bool is_volatile)
       EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
   void SetFlags(const std::set<RTCPPacketType>& types, bool is_volatile)
       EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
   bool IsFlagPresent(uint32_t type) const
       EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
@@ skipping 15 matching lines @@
   typedef std::unique_ptr<rtcp::RtcpPacket> (RTCPSender::*BuilderFunc)(
       const RtcpContext&);
   // Map from RTCPPacketType to builder.
   std::map<uint32_t, BuilderFunc> builders_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTCPSender);
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_SENDER_H_