[rtp_rtcp] lint errors about rand() usage fixed.

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.
  */
 
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_SENDER_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_SENDER_H_
 
 #include <map>
 #include <set>
 #include <sstream>
 #include <string>
 #include <vector>
 
+#include "webrtc/base/random.h"
 #include "webrtc/base/scoped_ptr.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_packet.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/modules/rtp_rtcp/source/tmmbr_help.h"
@@ skipping 165 matching lines @@
       EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
   rtc::scoped_ptr<rtcp::RtcpPacket> BuildReceiverReferenceTime(
       const RtcpContext& context)
       EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
   rtc::scoped_ptr<rtcp::RtcpPacket> BuildDlrr(const RtcpContext& context)
       EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
 
  private:
   const bool audio_;
   Clock* const clock_;
+  Random random_ GUARDED_BY(critical_section_rtcp_sender_);
   RtcpMode method_ GUARDED_BY(critical_section_rtcp_sender_);
 
   Transport* const transport_;
 
   rtc::scoped_ptr<CriticalSectionWrapper> critical_section_rtcp_sender_;
   bool using_nack_ GUARDED_BY(critical_section_rtcp_sender_);
   bool sending_ GUARDED_BY(critical_section_rtcp_sender_);
   bool remb_enabled_ GUARDED_BY(critical_section_rtcp_sender_);
 
   int64_t next_time_to_send_rtcp_ GUARDED_BY(critical_section_rtcp_sender_);
@@ skipping 79 matching lines @@
 
   std::set<ReportFlag> report_flags_ GUARDED_BY(critical_section_rtcp_sender_);
 
   typedef rtc::scoped_ptr<rtcp::RtcpPacket> (RTCPSender::*BuilderFunc)(
       const RtcpContext&);
   std::map<RTCPPacketType, BuilderFunc> builders_;
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_SENDER_H_