Remove mutable from rtc::CriticalSection members.

webrtc/modules/audio_processing/audio_processing_performance_unittest.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 "webrtc/modules/audio_processing/audio_processing_impl.h"
@@ skipping 184 matching lines @@
   int RenderMinusCaptureCounters() const {
     return -CaptureMinusRenderCounters();
   }
 
   bool BothCountersExceedeThreshold(int threshold) const {
     rtc::CritScope cs(&crit_);
     return (render_count_ > threshold && capture_count_ > threshold);
   }
 
  private:
-  mutable rtc::CriticalSection crit_;
+  rtc::CriticalSection crit_;
   int render_count_ GUARDED_BY(crit_) = 0;
   int capture_count_ GUARDED_BY(crit_) = 0;
 };
 
 // Class that protects a flag using a lock.
 class LockedFlag {
  public:
   bool get_flag() const {
     rtc::CritScope cs(&crit_);
     return flag_;
   }
 
   void set_flag() {
     rtc::CritScope cs(&crit_);
     flag_ = true;
   }
 
  private:
-  mutable rtc::CriticalSection crit_;
+  rtc::CriticalSection crit_;
   bool flag_ GUARDED_BY(crit_) = false;
 };
 
 // Parent class for the thread processors.
 class TimedThreadApiProcessor {
  public:
   TimedThreadApiProcessor(ProcessorType processor_type,
                           Random* rand_gen,
                           FrameCounters* shared_counters_state,
                           LockedFlag* capture_call_checker,
@@ skipping 480 matching lines @@
   // Run test and verify that it did not time out.
   EXPECT_EQ(kEventSignaled, Run());
 }
 
 INSTANTIATE_TEST_CASE_P(
     AudioProcessingPerformanceTest,
     CallSimulator,
     ::testing::ValuesIn(SimulationConfig::GenerateSimulationConfigs()));
 
 }  // namespace webrtc