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

webrtc/modules/rtp_rtcp/source/rtcp_sender.cc

 /*
  *  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.
  */
 
 #include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
 
 #include <assert.h>  // assert
-#include <stdlib.h>  // rand
 #include <string.h>  // memcpy
 
 #include <algorithm>  // min
 #include <limits>     // max
 #include <utility>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/common_types.h"
@@ skipping 109 matching lines @@
 };
 
 RTCPSender::RTCPSender(
     bool audio,
     Clock* clock,
     ReceiveStatistics* receive_statistics,
     RtcpPacketTypeCounterObserver* packet_type_counter_observer,
     Transport* outgoing_transport)
     : audio_(audio),
       clock_(clock),
+      random_(clock_->TimeInMicroseconds()),
       method_(RtcpMode::kOff),
       transport_(outgoing_transport),
 
       critical_section_rtcp_sender_(
           CriticalSectionWrapper::CreateCriticalSection()),
       using_nack_(false),
       sending_(false),
       remb_enabled_(false),
       next_time_to_send_rtcp_(0),
       start_timestamp_(0),
@@ skipping 753 matching lines @@
   if (IsFlagPresent(kRtcpSr) || (IsFlagPresent(kRtcpRr) && !cname_.empty()))
     SetFlag(kRtcpSdes, true);
 
   if (generate_report) {
     if (!sending_ && xr_send_receiver_reference_time_enabled_)
       SetFlag(kRtcpXrReceiverReferenceTime, true);
     if (feedback_state.has_last_xr_rr)
       SetFlag(kRtcpXrDlrrReportBlock, true);
 
     // generate next time to send an RTCP report
-    // seeded from RTP constructor
-    int32_t random = rand() % 1000;
-    int32_t timeToNext = RTCP_INTERVAL_AUDIO_MS;
+    uint32_t minIntervalMs = RTCP_INTERVAL_AUDIO_MS;
 
-    if (audio_) {
-      timeToNext = (RTCP_INTERVAL_AUDIO_MS / 2) +
-                   (RTCP_INTERVAL_AUDIO_MS * random / 1000);
-    } else {
-      uint32_t minIntervalMs = RTCP_INTERVAL_AUDIO_MS;
+    if (!audio_) {
       if (sending_) {
         // Calculate bandwidth for video; 360 / send bandwidth in kbit/s.
         uint32_t send_bitrate_kbit = feedback_state.send_bitrate / 1000;
         if (send_bitrate_kbit != 0)
           minIntervalMs = 360000 / send_bitrate_kbit;
       }
       if (minIntervalMs > RTCP_INTERVAL_VIDEO_MS)
         minIntervalMs = RTCP_INTERVAL_VIDEO_MS;
-      timeToNext = (minIntervalMs / 2) + (minIntervalMs * random / 1000);
     }
+    // The interval between RTCP packets is varied randomly over the
+    // range [1/2,3/2] times the calculated interval.
+    uint32_t timeToNext =
+        random_.Rand(minIntervalMs * 1 / 2, minIntervalMs * 3 / 2);
     next_time_to_send_rtcp_ = clock_->TimeInMilliseconds() + timeToNext;
 
     StatisticianMap statisticians =
         receive_statistics_->GetActiveStatisticians();
     if (!statisticians.empty()) {
       for (auto it = statisticians.begin(); it != statisticians.end(); ++it) {
         RTCPReportBlock report_block;
         if (PrepareReportBlock(feedback_state, it->first, it->second,
                                &report_block)) {
           // TODO(danilchap) AddReportBlock may fail (for 2 different reasons).
@@ skipping 145 matching lines @@
     Transport* const transport_;
     bool send_failure_;
   } sender(transport_);
 
   uint8_t buffer[IP_PACKET_SIZE];
   return packet.BuildExternalBuffer(buffer, IP_PACKET_SIZE, &sender) &&
          !sender.send_failure_;
 }
 
 }  // namespace webrtc