Add SafeClamp(), which accepts args of different types

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.
  */
 
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h"
 
 #include <stdio.h>
 
 #include <sstream>
 
 #include "webrtc/base/constructormagic.h"
+#include "webrtc/base/safe_minmax.h"
 
 namespace webrtc {
 namespace testing {
 namespace bwe {
 
 class DelayCapHelper {
  public:
   // Max delay = 0 stands for +infinite.
   DelayCapHelper() : max_delay_us_(0), delay_stats_() {}
 
@@ skipping 363 matching lines @@
   BWE_TEST_LOGGING_LOG1("Max Jitter", "%d ms", static_cast<int>(max_jitter_ms));
   assert(max_jitter_ms >= 0);
   // Truncated gaussian, Max jitter = kN*sigma.
   stddev_jitter_us_ = (max_jitter_ms * 1000 + kN / 2) / kN;
 }
 
 namespace {
 inline int64_t TruncatedNSigmaGaussian(Random* const random,
                                        int64_t mean,
                                        int64_t std_dev) {
-  int64_t gaussian_random = random->Gaussian(mean, std_dev);
-  return std::max(std::min(gaussian_random, kN * std_dev), -kN * std_dev);
+  const int64_t gaussian_random = random->Gaussian(mean, std_dev);
+  return rtc::SafeClamp(-kN * std_dev, kN * std_dev, gaussian_random);
 }
 }
 
 void JitterFilter::RunFor(int64_t /*time_ms*/, Packets* in_out) {
   assert(in_out);
   for (Packet* packet : *in_out) {
     int64_t jitter_us =
         std::abs(TruncatedNSigmaGaussian(&random_, 0, stddev_jitter_us_));
     int64_t new_send_time_us = packet->send_time_us() + jitter_us;
 
@@ skipping 398 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