Use RTC_UNUSED instead of conditional compilation in BWE simulator tool.

webrtc/modules/remote_bitrate_estimator/test/estimators/remb.cc

 /*
  *  Copyright (c) 2015 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 <algorithm>
 
 #include "webrtc/modules/remote_bitrate_estimator/test/estimators/remb.h"
 
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
 #include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h"
 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
 #include "webrtc/test/gtest.h"
+#include "webrtc/typedefs.h"
 
 namespace webrtc {
 namespace testing {
 namespace bwe {
 
 RembBweSender::RembBweSender(int kbps, BitrateObserver* observer, Clock* clock)
     : bitrate_controller_(
           BitrateController::CreateBitrateController(clock,
                                                      observer,
                                                      &event_log_)),
@@ skipping 28 matching lines @@
   bitrate_controller_->Process();
 }
 
 int RembBweSender::GetFeedbackIntervalMs() const {
   return 100;
 }
 
 RembReceiver::RembReceiver(int flow_id, bool plot)
     : BweReceiver(flow_id),
       estimate_log_prefix_(),
-#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
       plot_estimate_(plot),
-#endif
       clock_(0),
       recv_stats_(ReceiveStatistics::Create(&clock_)),
       latest_estimate_bps_(-1),
       last_feedback_ms_(-1),
       estimator_(new RemoteBitrateEstimatorAbsSendTime(this, &clock_)) {
   std::stringstream ss;
   ss << "Estimate_" << flow_id_ << "#1";
   estimate_log_prefix_ = ss.str();
   // Default RTT in RemoteRateControl is 200 ms ; 50 ms is more realistic.
   estimator_->OnRttUpdate(50, 50);
@@ skipping 33 matching lines @@
     StatisticianMap statisticians = recv_stats_->GetActiveStatisticians();
     RTCPReportBlock report_block;
     if (!statisticians.empty()) {
       latest_report_block_ = BuildReportBlock(statisticians.begin()->second);
     }
 
     feedback = new RembFeedback(flow_id_, now_ms * 1000, last_feedback_ms_,
                                 estimated_bps, latest_report_block_);
     last_feedback_ms_ = now_ms;
 
-#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
     double estimated_kbps = static_cast<double>(estimated_bps) / 1000.0;
+    RTC_UNUSED(estimated_kbps);
     if (plot_estimate_) {
       BWE_TEST_LOGGING_PLOT(0, estimate_log_prefix_,
                             clock_.TimeInMilliseconds(), estimated_kbps);
     }
-#endif
   }
   return feedback;
 }
 
 void RembReceiver::OnReceiveBitrateChanged(const std::vector<uint32_t>& ssrcs,
                                            uint32_t bitrate) {}
 
 RTCPReportBlock RembReceiver::BuildReportBlock(
     StreamStatistician* statistician) {
   RTCPReportBlock report_block;
@@ skipping 15 matching lines @@
     }
     latest_estimate_bps_ = bps;
   }
   *estimate_bps = latest_estimate_bps_;
   return true;
 }
 
 }  // namespace bwe
 }  // namespace testing
 }  // namespace webrtc