Add SafeClamp(), which accepts args of different types

webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc

 /*
  *  Copyright (c) 2014 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/audio_coding/codecs/opus/audio_encoder_opus.h"
 
 #include <algorithm>
 #include <iterator>
 #include <utility>
 
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/numerics/exp_filter.h"
 #include "webrtc/base/protobuf_utils.h"
 #include "webrtc/base/safe_conversions.h"
+#include "webrtc/base/safe_minmax.h"
 #include "webrtc/base/string_to_number.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.h"
 #include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
 #include "webrtc/system_wrappers/include/field_trial.h"
 
 namespace webrtc {
 
@@ skipping 650 matching lines @@
   float opt_loss_rate = OptimizePacketLossRate(fraction, packet_loss_rate_);
   if (packet_loss_rate_ != opt_loss_rate) {
     packet_loss_rate_ = opt_loss_rate;
     RTC_CHECK_EQ(
         0, WebRtcOpus_SetPacketLossRate(
                inst_, static_cast<int32_t>(packet_loss_rate_ * 100 + .5)));
   }
 }
 
 void AudioEncoderOpus::SetTargetBitrate(int bits_per_second) {
-  config_.bitrate_bps = rtc::Optional<int>(std::max(
-      std::min(bits_per_second, kOpusMaxBitrateBps), kOpusMinBitrateBps));
+  config_.bitrate_bps = rtc::Optional<int>(rtc::SafeClamp<int>(
+      bits_per_second, kOpusMinBitrateBps, kOpusMaxBitrateBps));
   RTC_DCHECK(config_.IsOk());
   RTC_CHECK_EQ(0, WebRtcOpus_SetBitRate(inst_, config_.GetBitrateBps()));
   const auto new_complexity = config_.GetNewComplexity();
   if (new_complexity && complexity_ != *new_complexity) {
     complexity_ = *new_complexity;
     RTC_CHECK_EQ(0, WebRtcOpus_SetComplexity(inst_, complexity_));
   }
 }
 
 void AudioEncoderOpus::ApplyAudioNetworkAdaptor() {
@@ skipping 37 matching lines @@
             config_.uplink_bandwidth_update_interval_ms) {
       rtc::Optional<float> smoothed_bitrate = bitrate_smoother_->GetAverage();
       if (smoothed_bitrate)
         audio_network_adaptor_->SetUplinkBandwidth(*smoothed_bitrate);
       bitrate_smoother_last_update_time_ = rtc::Optional<int64_t>(now_ms);
     }
   }
 }
 
 }  // namespace webrtc