Fixed lint warnings in webrtc/modules/remote_bitrate_estimator

webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.cc

 /*
  *  Copyright (c) 2013 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 568 matching lines @@
 
 TraceBasedDeliveryFilter::~TraceBasedDeliveryFilter() {
 }
 
 bool TraceBasedDeliveryFilter::Init(const std::string& filename) {
   FILE* trace_file = fopen(filename.c_str(), "r");
   if (!trace_file) {
     return false;
   }
   int64_t first_timestamp = -1;
-  while(!feof(trace_file)) {
+  while (!feof(trace_file)) {
     const size_t kMaxLineLength = 100;
     char line[kMaxLineLength];
     if (fgets(line, kMaxLineLength, trace_file)) {
       std::string line_string(line);
       std::istringstream buffer(line_string);
       int64_t timestamp;
       buffer >> timestamp;
       timestamp /= 1000;  // Convert to microseconds.
       if (first_timestamp == -1)
         first_timestamp = timestamp;
@@ skipping 73 matching lines @@
 VideoSource::VideoSource(int flow_id,
                          float fps,
                          uint32_t kbps,
                          uint32_t ssrc,
                          int64_t first_frame_offset_ms)
     : kMaxPayloadSizeBytes(1200),
       kTimestampBase(0xff80ff00ul),
       frame_period_ms_(1000.0 / fps),
       bits_per_second_(1000 * kbps),
       frame_size_bytes_(bits_per_second_ / 8 / fps),
+      random_(0x12345678),
       flow_id_(flow_id),
       next_frame_ms_(first_frame_offset_ms),
       next_frame_rand_ms_(0),
       now_ms_(0),
       prototype_header_() {
   memset(&prototype_header_, 0, sizeof(prototype_header_));
   prototype_header_.ssrc = ssrc;
   prototype_header_.sequenceNumber = 0xf000u;
 }
 
@@ skipping 13 matching lines @@
 void VideoSource::RunFor(int64_t time_ms, Packets* in_out) {
   assert(in_out);
 
   now_ms_ += time_ms;
   Packets new_packets;
 
   while (now_ms_ >= next_frame_ms_) {
     const int64_t kRandAmplitude = 2;
     // A variance picked uniformly from {-1, 0, 1} ms is added to the frame
     // timestamp.
-    next_frame_rand_ms_ =
-        kRandAmplitude * static_cast<float>(rand()) / RAND_MAX -
-        kRandAmplitude / 2;
+    next_frame_rand_ms_ = kRandAmplitude * (random_.Rand<float>() - 0.5);
 
     // Ensure frame will not have a negative timestamp.
     int64_t next_frame_ms =
         std::max<int64_t>(next_frame_ms_ + next_frame_rand_ms_, 0);
 
     prototype_header_.timestamp =
         kTimestampBase + static_cast<uint32_t>(next_frame_ms * 90.0);
     prototype_header_.extension.transmissionTimeOffset = 0;
 
     // Generate new packets for this frame, all with the same timestamp,
@@ skipping 79 matching lines @@
 uint32_t PeriodicKeyFrameSource::NextPacketSize(uint32_t frame_size,
                                                 uint32_t remaining_payload) {
   uint32_t fragments =
       (frame_size + (kMaxPayloadSizeBytes - 1)) / kMaxPayloadSizeBytes;
   uint32_t avg_size = (frame_size + fragments - 1) / fragments;
   return std::min(avg_size, remaining_payload);
 }
 }  // namespace bwe
 }  // namespace testing
 }  // namespace webrtc