diff --git a/webrtc/modules/audio_conference_mixer/source/time_scheduler.cc b/webrtc/modules/audio_conference_mixer/source/time_scheduler.cc
index 30b2933b61..877d98b053 100644
--- a/webrtc/modules/audio_conference_mixer/source/time_scheduler.cc
+++ b/webrtc/modules/audio_conference_mixer/source/time_scheduler.cc
@@ -10,27 +10,17 @@
 
 #include "webrtc/base/timeutils.h"
 #include "webrtc/modules/audio_conference_mixer/source/time_scheduler.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
 TimeScheduler::TimeScheduler(const int64_t periodicityInMs)
-    : _crit(CriticalSectionWrapper::CreateCriticalSection()),
-      _isStarted(false),
+    : _isStarted(false),
       _lastPeriodMark(),
       _periodicityInMs(periodicityInMs),
       _periodicityInTicks(periodicityInMs * rtc::kNumNanosecsPerMillisec),
-      _missedPeriods(0)
- {
- }
+      _missedPeriods(0) {}
 
-TimeScheduler::~TimeScheduler()
-{
-    delete _crit;
-}
-
-int32_t TimeScheduler::UpdateScheduler()
-{
-    CriticalSectionScoped cs(_crit);
+int32_t TimeScheduler::UpdateScheduler() {
+    rtc::CritScope cs(&_crit);
     if(!_isStarted)
     {
         _isStarted = true;
@@ -79,7 +69,7 @@ int32_t TimeScheduler::UpdateScheduler()
 int32_t TimeScheduler::TimeToNextUpdate(
     int64_t& updateTimeInMS) const
 {
-    CriticalSectionScoped cs(_crit);
+    rtc::CritScope cs(&_crit);
     // Missed periods means that the next UpdateScheduler() should happen
     // immediately.
     if(_missedPeriods > 0)
diff --git a/webrtc/modules/audio_conference_mixer/source/time_scheduler.h b/webrtc/modules/audio_conference_mixer/source/time_scheduler.h
index d1897fa100..b155429b8b 100644
--- a/webrtc/modules/audio_conference_mixer/source/time_scheduler.h
+++ b/webrtc/modules/audio_conference_mixer/source/time_scheduler.h
@@ -11,17 +11,18 @@
 // The TimeScheduler class keeps track of periodic events. It is non-drifting
 // and keeps track of any missed periods so that it is possible to catch up.
 // (compare to a metronome)
+#include "webrtc/base/criticalsection.h"
 
 #ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_TIME_SCHEDULER_H_
 #define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_TIME_SCHEDULER_H_
 
 namespace webrtc {
 class CriticalSectionWrapper;
-class TimeScheduler
-{
+
+class TimeScheduler {
 public:
     TimeScheduler(const int64_t periodicityInMs);
-    ~TimeScheduler();
+    ~TimeScheduler() = default;
 
     // Signal that a periodic event has been triggered.
     int32_t UpdateScheduler();
@@ -31,7 +32,7 @@ public:
     int32_t TimeToNextUpdate(int64_t& updateTimeInMS) const;
 
 private:
-    CriticalSectionWrapper* _crit;
+    rtc::CriticalSection _crit;
 
     bool _isStarted;
     int64_t _lastPeriodMark; // In ns
diff --git a/webrtc/modules/audio_device/audio_device_impl.cc b/webrtc/modules/audio_device/audio_device_impl.cc
index 5040189eeb..45bc3a5aaf 100644
--- a/webrtc/modules/audio_device/audio_device_impl.cc
+++ b/webrtc/modules/audio_device/audio_device_impl.cc
@@ -53,7 +53,6 @@
 
 #include "webrtc/modules/audio_device/dummy/audio_device_dummy.h"
 #include "webrtc/modules/audio_device/dummy/file_audio_device.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 #define CHECK_INITIALIZED() \
   {                         \
@@ -118,10 +117,7 @@ rtc::scoped_refptr<AudioDeviceModule> AudioDeviceModule::Create(
 
 AudioDeviceModuleImpl::AudioDeviceModuleImpl(const int32_t id,
                                              const AudioLayer audioLayer)
-    : _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
-      _critSectEventCb(*CriticalSectionWrapper::CreateCriticalSection()),
-      _critSectAudioCb(*CriticalSectionWrapper::CreateCriticalSection()),
-      _ptrCbAudioDeviceObserver(NULL),
+    : _ptrCbAudioDeviceObserver(NULL),
       _ptrAudioDevice(NULL),
       _id(id),
       _platformAudioLayer(audioLayer),
@@ -358,15 +354,10 @@ int32_t AudioDeviceModuleImpl::AttachAudioBuffer() {
 
 AudioDeviceModuleImpl::~AudioDeviceModuleImpl() {
   LOG(INFO) << __FUNCTION__;
-
   if (_ptrAudioDevice) {
     delete _ptrAudioDevice;
     _ptrAudioDevice = NULL;
   }
-
-  delete &_critSect;
-  delete &_critSectEventCb;
-  delete &_critSectAudioCb;
 }
 
 // ============================================================================
@@ -398,7 +389,7 @@ void AudioDeviceModuleImpl::Process() {
 
   // kPlayoutWarning
   if (_ptrAudioDevice->PlayoutWarning()) {
-    CriticalSectionScoped lock(&_critSectEventCb);
+    rtc::CritScope lock(&_critSectEventCb);
     if (_ptrCbAudioDeviceObserver) {
       LOG(WARNING) << "=> OnWarningIsReported(kPlayoutWarning)";
       _ptrCbAudioDeviceObserver->OnWarningIsReported(
@@ -409,7 +400,7 @@ void AudioDeviceModuleImpl::Process() {
 
   // kPlayoutError
   if (_ptrAudioDevice->PlayoutError()) {
-    CriticalSectionScoped lock(&_critSectEventCb);
+    rtc::CritScope lock(&_critSectEventCb);
     if (_ptrCbAudioDeviceObserver) {
       LOG(LERROR) << "=> OnErrorIsReported(kPlayoutError)";
       _ptrCbAudioDeviceObserver->OnErrorIsReported(
@@ -420,7 +411,7 @@ void AudioDeviceModuleImpl::Process() {
 
   // kRecordingWarning
   if (_ptrAudioDevice->RecordingWarning()) {
-    CriticalSectionScoped lock(&_critSectEventCb);
+    rtc::CritScope lock(&_critSectEventCb);
     if (_ptrCbAudioDeviceObserver) {
       LOG(WARNING) << "=> OnWarningIsReported(kRecordingWarning)";
       _ptrCbAudioDeviceObserver->OnWarningIsReported(
@@ -431,7 +422,7 @@ void AudioDeviceModuleImpl::Process() {
 
   // kRecordingError
   if (_ptrAudioDevice->RecordingError()) {
-    CriticalSectionScoped lock(&_critSectEventCb);
+    rtc::CritScope lock(&_critSectEventCb);
     if (_ptrCbAudioDeviceObserver) {
       LOG(LERROR) << "=> OnErrorIsReported(kRecordingError)";
       _ptrCbAudioDeviceObserver->OnErrorIsReported(
@@ -1459,7 +1450,7 @@ bool AudioDeviceModuleImpl::Recording() const {
 int32_t AudioDeviceModuleImpl::RegisterEventObserver(
     AudioDeviceObserver* eventCallback) {
   LOG(INFO) << __FUNCTION__;
-  CriticalSectionScoped lock(&_critSectEventCb);
+  rtc::CritScope lock(&_critSectEventCb);
   _ptrCbAudioDeviceObserver = eventCallback;
 
   return 0;
@@ -1472,7 +1463,7 @@ int32_t AudioDeviceModuleImpl::RegisterEventObserver(
 int32_t AudioDeviceModuleImpl::RegisterAudioCallback(
     AudioTransport* audioCallback) {
   LOG(INFO) << __FUNCTION__;
-  CriticalSectionScoped lock(&_critSectAudioCb);
+  rtc::CritScope lock(&_critSectAudioCb);
   return _audioDeviceBuffer.RegisterAudioCallback(audioCallback);
 }
 
diff --git a/webrtc/modules/audio_device/audio_device_impl.h b/webrtc/modules/audio_device/audio_device_impl.h
index 034690f6b2..709eb769bd 100644
--- a/webrtc/modules/audio_device/audio_device_impl.h
+++ b/webrtc/modules/audio_device/audio_device_impl.h
@@ -16,6 +16,7 @@
 #include <memory>
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/modules/audio_device/audio_device_buffer.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
 
@@ -23,7 +24,6 @@ namespace webrtc {
 
 class AudioDeviceGeneric;
 class AudioManager;
-class CriticalSectionWrapper;
 
 class AudioDeviceModuleImpl : public AudioDeviceModule {
  public:
@@ -203,9 +203,9 @@ class AudioDeviceModuleImpl : public AudioDeviceModule {
   PlatformType Platform() const;
   AudioLayer PlatformAudioLayer() const;
 
-  CriticalSectionWrapper& _critSect;
-  CriticalSectionWrapper& _critSectEventCb;
-  CriticalSectionWrapper& _critSectAudioCb;
+  rtc::CriticalSection _critSect;
+  rtc::CriticalSection _critSectEventCb;
+  rtc::CriticalSection _critSectAudioCb;
 
   AudioDeviceObserver* _ptrCbAudioDeviceObserver;
 
diff --git a/webrtc/modules/audio_device/dummy/file_audio_device.cc b/webrtc/modules/audio_device/dummy/file_audio_device.cc
index 3dcb1384e7..92b09e0a9d 100644
--- a/webrtc/modules/audio_device/dummy/file_audio_device.cc
+++ b/webrtc/modules/audio_device/dummy/file_audio_device.cc
@@ -33,7 +33,6 @@ FileAudioDevice::FileAudioDevice(const int32_t id,
     _playoutBuffer(NULL),
     _recordingFramesLeft(0),
     _playoutFramesLeft(0),
-    _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
     _recordingBufferSizeIn10MS(0),
     _recordingFramesIn10MS(0),
     _playoutFramesIn10MS(0),
@@ -161,7 +160,7 @@ int32_t FileAudioDevice::RecordingIsAvailable(bool& available) {
 }
 
 int32_t FileAudioDevice::InitRecording() {
-  CriticalSectionScoped lock(&_critSect);
+  rtc::CritScope lock(&_critSect);
 
   if (_recording) {
     return -1;
@@ -219,7 +218,7 @@ int32_t FileAudioDevice::StartPlayout() {
 
 int32_t FileAudioDevice::StopPlayout() {
   {
-      CriticalSectionScoped lock(&_critSect);
+      rtc::CritScope lock(&_critSect);
       _playing = false;
   }
 
@@ -229,7 +228,7 @@ int32_t FileAudioDevice::StopPlayout() {
       _ptrThreadPlay.reset();
   }
 
-  CriticalSectionScoped lock(&_critSect);
+  rtc::CritScope lock(&_critSect);
 
   _playoutFramesLeft = 0;
   delete [] _playoutBuffer;
@@ -280,7 +279,7 @@ int32_t FileAudioDevice::StartRecording() {
 
 int32_t FileAudioDevice::StopRecording() {
   {
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
     _recording = false;
   }
 
@@ -289,7 +288,7 @@ int32_t FileAudioDevice::StopRecording() {
       _ptrThreadRec.reset();
   }
 
-  CriticalSectionScoped lock(&_critSect);
+  rtc::CritScope lock(&_critSect);
   _recordingFramesLeft = 0;
   if (_recordingBuffer) {
       delete [] _recordingBuffer;
@@ -456,7 +455,7 @@ void FileAudioDevice::ClearRecordingWarning() {}
 void FileAudioDevice::ClearRecordingError() {}
 
 void FileAudioDevice::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
-  CriticalSectionScoped lock(&_critSect);
+  rtc::CritScope lock(&_critSect);
 
   _ptrAudioBuffer = audioBuffer;
 
diff --git a/webrtc/modules/audio_device/dummy/file_audio_device.h b/webrtc/modules/audio_device/dummy/file_audio_device.h
index 2065f21d47..41760552c3 100644
--- a/webrtc/modules/audio_device/dummy/file_audio_device.h
+++ b/webrtc/modules/audio_device/dummy/file_audio_device.h
@@ -16,9 +16,9 @@
 #include <memory>
 #include <string>
 
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/modules/audio_device/audio_device_generic.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/file_wrapper.h"
 
 namespace rtc {
@@ -176,7 +176,7 @@ class FileAudioDevice : public AudioDeviceGeneric {
   int8_t* _playoutBuffer;  // In bytes.
   uint32_t _recordingFramesLeft;
   uint32_t _playoutFramesLeft;
-  CriticalSectionWrapper& _critSect;
+  rtc::CriticalSection _critSect;
 
   size_t _recordingBufferSizeIn10MS;
   size_t _recordingFramesIn10MS;
diff --git a/webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc b/webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc
index f7315de2af..3b8cf24773 100644
--- a/webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc
+++ b/webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc
@@ -62,7 +62,6 @@ static const unsigned int ALSA_CAPTURE_WAIT_TIMEOUT = 5; // in ms
 
 AudioDeviceLinuxALSA::AudioDeviceLinuxALSA(const int32_t id) :
     _ptrAudioBuffer(NULL),
-    _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
     _id(id),
     _mixerManager(id),
     _inputDeviceIndex(0),
@@ -130,13 +129,12 @@ AudioDeviceLinuxALSA::~AudioDeviceLinuxALSA()
         delete [] _playoutBuffer;
         _playoutBuffer = NULL;
     }
-    delete &_critSect;
 }
 
 void AudioDeviceLinuxALSA::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer)
 {
 
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
 
     _ptrAudioBuffer = audioBuffer;
 
@@ -157,7 +155,7 @@ int32_t AudioDeviceLinuxALSA::ActiveAudioLayer(
 }
 
 AudioDeviceGeneric::InitStatus AudioDeviceLinuxALSA::Init() {
-  CriticalSectionScoped lock(&_critSect);
+  rtc::CritScope lock(&_critSect);
 
   // Load libasound
   if (!AlsaSymbolTable.Load()) {
@@ -194,7 +192,7 @@ int32_t AudioDeviceLinuxALSA::Terminate()
         return 0;
     }
 
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
 
     _mixerManager.Close();
 
@@ -243,7 +241,7 @@ bool AudioDeviceLinuxALSA::Initialized() const
 int32_t AudioDeviceLinuxALSA::InitSpeaker()
 {
 
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
 
     if (_playing)
     {
@@ -258,7 +256,7 @@ int32_t AudioDeviceLinuxALSA::InitSpeaker()
 int32_t AudioDeviceLinuxALSA::InitMicrophone()
 {
 
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
 
     if (_recording)
     {
@@ -560,7 +558,7 @@ int32_t AudioDeviceLinuxALSA::MicrophoneBoost(bool& enabled) const
 int32_t AudioDeviceLinuxALSA::StereoRecordingIsAvailable(bool& available)
 {
 
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
 
     // If we already have initialized in stereo it's obviously available
     if (_recIsInitialized && (2 == _recChannels))
@@ -631,7 +629,7 @@ int32_t AudioDeviceLinuxALSA::StereoRecording(bool& enabled) const
 int32_t AudioDeviceLinuxALSA::StereoPlayoutIsAvailable(bool& available)
 {
 
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
 
     // If we already have initialized in stereo it's obviously available
     if (_playIsInitialized && (2 == _playChannels))
@@ -1009,7 +1007,7 @@ int32_t AudioDeviceLinuxALSA::InitPlayout()
 
     int errVal = 0;
 
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
     if (_playing)
     {
         return -1;
@@ -1162,7 +1160,7 @@ int32_t AudioDeviceLinuxALSA::InitRecording()
 
     int errVal = 0;
 
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
 
     if (_recording)
     {
@@ -1397,7 +1395,7 @@ int32_t AudioDeviceLinuxALSA::StopRecording()
 {
 
     {
-      CriticalSectionScoped lock(&_critSect);
+      rtc::CritScope lock(&_critSect);
 
       if (!_recIsInitialized)
       {
@@ -1420,7 +1418,7 @@ int32_t AudioDeviceLinuxALSA::StopRecording()
         _ptrThreadRec.reset();
     }
 
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
     _recordingFramesLeft = 0;
     if (_recordingBuffer)
     {
@@ -1523,7 +1521,7 @@ int32_t AudioDeviceLinuxALSA::StopPlayout()
 {
 
     {
-        CriticalSectionScoped lock(&_critSect);
+        rtc::CritScope lock(&_critSect);
 
         if (!_playIsInitialized)
         {
@@ -1545,7 +1543,7 @@ int32_t AudioDeviceLinuxALSA::StopPlayout()
         _ptrThreadPlay.reset();
     }
 
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
 
     _playoutFramesLeft = 0;
     delete [] _playoutBuffer;
@@ -1635,49 +1633,49 @@ int32_t AudioDeviceLinuxALSA::CPULoad(uint16_t& load) const
 
 bool AudioDeviceLinuxALSA::PlayoutWarning() const
 {
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
     return (_playWarning > 0);
 }
 
 bool AudioDeviceLinuxALSA::PlayoutError() const
 {
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
     return (_playError > 0);
 }
 
 bool AudioDeviceLinuxALSA::RecordingWarning() const
 {
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
     return (_recWarning > 0);
 }
 
 bool AudioDeviceLinuxALSA::RecordingError() const
 {
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
     return (_recError > 0);
 }
 
 void AudioDeviceLinuxALSA::ClearPlayoutWarning()
 {
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
     _playWarning = 0;
 }
 
 void AudioDeviceLinuxALSA::ClearPlayoutError()
 {
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
     _playError = 0;
 }
 
 void AudioDeviceLinuxALSA::ClearRecordingWarning()
 {
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
     _recWarning = 0;
 }
 
 void AudioDeviceLinuxALSA::ClearRecordingError()
 {
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
     _recError = 0;
 }
 
diff --git a/webrtc/modules/audio_device/linux/audio_device_alsa_linux.h b/webrtc/modules/audio_device/linux/audio_device_alsa_linux.h
index aba62e8669..40525982bb 100644
--- a/webrtc/modules/audio_device/linux/audio_device_alsa_linux.h
+++ b/webrtc/modules/audio_device/linux/audio_device_alsa_linux.h
@@ -13,10 +13,10 @@
 
 #include <memory>
 
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/base/platform_thread.h"
 #include "webrtc/modules/audio_device/audio_device_generic.h"
 #include "webrtc/modules/audio_device/linux/audio_mixer_manager_alsa_linux.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 #if defined(USE_X11)
 #include <X11/Xlib.h>
@@ -145,7 +145,6 @@ public:
     // CPU load
     int32_t CPULoad(uint16_t& load) const override;
 
-public:
  bool PlayoutWarning() const override;
  bool PlayoutError() const override;
  bool RecordingWarning() const override;
@@ -155,7 +154,6 @@ public:
  void ClearRecordingWarning() override;
  void ClearRecordingError() override;
 
-public:
  void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
 
 private:
@@ -166,26 +164,22 @@ private:
                            const int32_t ednLen = 0) const;
     int32_t ErrorRecovery(int32_t error, snd_pcm_t* deviceHandle);
 
-private:
     bool KeyPressed() const;
 
-private:
     void Lock() EXCLUSIVE_LOCK_FUNCTION(_critSect) { _critSect.Enter(); };
     void UnLock() UNLOCK_FUNCTION(_critSect) { _critSect.Leave(); };
-private:
+
     inline int32_t InputSanityCheckAfterUnlockedPeriod() const;
     inline int32_t OutputSanityCheckAfterUnlockedPeriod() const;
 
-private:
     static bool RecThreadFunc(void*);
     static bool PlayThreadFunc(void*);
     bool RecThreadProcess();
     bool PlayThreadProcess();
 
-private:
     AudioDeviceBuffer* _ptrAudioBuffer;
 
-    CriticalSectionWrapper& _critSect;
+    rtc::CriticalSection _critSect;
 
     // TODO(pbos): Make plain members and start/stop instead of resetting these
     // pointers. A thread can be reused.
@@ -226,7 +220,6 @@ private:
 
     AudioDeviceModule::BufferType _playBufType;
 
-private:
     bool _initialized;
     bool _recording;
     bool _playing;
diff --git a/webrtc/modules/audio_device/linux/audio_device_pulse_linux.cc b/webrtc/modules/audio_device/linux/audio_device_pulse_linux.cc
index e408f22066..a486ee5468 100644
--- a/webrtc/modules/audio_device/linux/audio_device_pulse_linux.cc
+++ b/webrtc/modules/audio_device/linux/audio_device_pulse_linux.cc
@@ -30,7 +30,6 @@ namespace webrtc
 
 AudioDeviceLinuxPulse::AudioDeviceLinuxPulse(const int32_t id) :
     _ptrAudioBuffer(NULL),
-    _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
     _timeEventRec(*EventWrapper::Create()),
     _timeEventPlay(*EventWrapper::Create()),
     _recStartEvent(*EventWrapper::Create()),
@@ -133,7 +132,6 @@ AudioDeviceLinuxPulse::~AudioDeviceLinuxPulse()
     delete &_playStartEvent;
     delete &_timeEventRec;
     delete &_timeEventPlay;
-    delete &_critSect;
 }
 
 void AudioDeviceLinuxPulse::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer)
@@ -751,7 +749,7 @@ int32_t AudioDeviceLinuxPulse::StereoPlayout(bool& enabled) const
 
 int32_t AudioDeviceLinuxPulse::SetAGC(bool enable)
 {
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
     _AGC = enable;
 
     return 0;
@@ -759,7 +757,7 @@ int32_t AudioDeviceLinuxPulse::SetAGC(bool enable)
 
 bool AudioDeviceLinuxPulse::AGC() const
 {
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
     return _AGC;
 }
 
@@ -1346,7 +1344,7 @@ int32_t AudioDeviceLinuxPulse::StartRecording()
     if (kEventTimeout == _recStartEvent.Wait(10000))
     {
         {
-            CriticalSectionScoped lock(&_critSect);
+            rtc::CritScope lock(&_critSect);
             _startRec = false;
         }
         StopRecording();
@@ -1356,7 +1354,7 @@ int32_t AudioDeviceLinuxPulse::StartRecording()
     }
 
     {
-        CriticalSectionScoped lock(&_critSect);
+        rtc::CritScope lock(&_critSect);
         if (_recording)
         {
             // The recording state is set by the audio thread after recording
@@ -1375,7 +1373,7 @@ int32_t AudioDeviceLinuxPulse::StartRecording()
 int32_t AudioDeviceLinuxPulse::StopRecording()
 {
     RTC_DCHECK(thread_checker_.CalledOnValidThread());
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
 
     if (!_recIsInitialized)
     {
@@ -1469,7 +1467,7 @@ int32_t AudioDeviceLinuxPulse::StartPlayout()
 
     // Set state to ensure that playout starts from the audio thread.
     {
-        CriticalSectionScoped lock(&_critSect);
+        rtc::CritScope lock(&_critSect);
         _startPlay = true;
     }
 
@@ -1481,7 +1479,7 @@ int32_t AudioDeviceLinuxPulse::StartPlayout()
     if (kEventTimeout == _playStartEvent.Wait(10000))
     {
         {
-            CriticalSectionScoped lock(&_critSect);
+            rtc::CritScope lock(&_critSect);
             _startPlay = false;
         }
         StopPlayout();
@@ -1491,7 +1489,7 @@ int32_t AudioDeviceLinuxPulse::StartPlayout()
     }
 
     {
-        CriticalSectionScoped lock(&_critSect);
+        rtc::CritScope lock(&_critSect);
         if (_playing)
         {
             // The playing state is set by the audio thread after playout
@@ -1510,7 +1508,7 @@ int32_t AudioDeviceLinuxPulse::StartPlayout()
 int32_t AudioDeviceLinuxPulse::StopPlayout()
 {
     RTC_DCHECK(thread_checker_.CalledOnValidThread());
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
 
     if (!_playIsInitialized)
     {
@@ -1574,7 +1572,7 @@ int32_t AudioDeviceLinuxPulse::StopPlayout()
 
 int32_t AudioDeviceLinuxPulse::PlayoutDelay(uint16_t& delayMS) const
 {
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
     delayMS = (uint16_t) _sndCardPlayDelay;
     return 0;
 }
@@ -1631,49 +1629,49 @@ int32_t AudioDeviceLinuxPulse::CPULoad(uint16_t& /*load*/) const
 
 bool AudioDeviceLinuxPulse::PlayoutWarning() const
 {
-  CriticalSectionScoped lock(&_critSect);
+  rtc::CritScope lock(&_critSect);
   return (_playWarning > 0);
 }
 
 bool AudioDeviceLinuxPulse::PlayoutError() const
 {
-  CriticalSectionScoped lock(&_critSect);
+  rtc::CritScope lock(&_critSect);
   return (_playError > 0);
 }
 
 bool AudioDeviceLinuxPulse::RecordingWarning() const
 {
-  CriticalSectionScoped lock(&_critSect);
+  rtc::CritScope lock(&_critSect);
   return (_recWarning > 0);
 }
 
 bool AudioDeviceLinuxPulse::RecordingError() const
 {
-  CriticalSectionScoped lock(&_critSect);
+  rtc::CritScope lock(&_critSect);
   return (_recError > 0);
 }
 
 void AudioDeviceLinuxPulse::ClearPlayoutWarning()
 {
-  CriticalSectionScoped lock(&_critSect);
+  rtc::CritScope lock(&_critSect);
   _playWarning = 0;
 }
 
 void AudioDeviceLinuxPulse::ClearPlayoutError()
 {
-  CriticalSectionScoped lock(&_critSect);
+  rtc::CritScope lock(&_critSect);
   _playError = 0;
 }
 
 void AudioDeviceLinuxPulse::ClearRecordingWarning()
 {
-  CriticalSectionScoped lock(&_critSect);
+  rtc::CritScope lock(&_critSect);
   _recWarning = 0;
 }
 
 void AudioDeviceLinuxPulse::ClearRecordingError()
 {
-  CriticalSectionScoped lock(&_critSect);
+  rtc::CritScope lock(&_critSect);
   _recError = 0;
 }
 
@@ -2602,7 +2600,7 @@ bool AudioDeviceLinuxPulse::PlayThreadProcess()
             return true;
     }
 
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
 
     if (_startPlay)
     {
@@ -2843,7 +2841,7 @@ bool AudioDeviceLinuxPulse::RecThreadProcess()
             return true;
     }
 
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
 
     if (_startRec)
     {
diff --git a/webrtc/modules/audio_device/linux/audio_device_pulse_linux.h b/webrtc/modules/audio_device/linux/audio_device_pulse_linux.h
index 3aa169807d..58934583db 100644
--- a/webrtc/modules/audio_device/linux/audio_device_pulse_linux.h
+++ b/webrtc/modules/audio_device/linux/audio_device_pulse_linux.h
@@ -13,11 +13,11 @@
 
 #include <memory>
 
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/base/platform_thread.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/modules/audio_device/audio_device_generic.h"
 #include "webrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 #include <X11/Xlib.h>
 #include <pulse/pulseaudio.h>
@@ -280,7 +280,7 @@ private:
 
     AudioDeviceBuffer* _ptrAudioBuffer;
 
-    CriticalSectionWrapper& _critSect;
+    rtc::CriticalSection _critSect;
     EventWrapper& _timeEventRec;
     EventWrapper& _timeEventPlay;
     EventWrapper& _recStartEvent;
diff --git a/webrtc/modules/audio_device/linux/audio_mixer_manager_alsa_linux.cc b/webrtc/modules/audio_device/linux/audio_mixer_manager_alsa_linux.cc
index 29620eb043..be5e17e14c 100644
--- a/webrtc/modules/audio_device/linux/audio_mixer_manager_alsa_linux.cc
+++ b/webrtc/modules/audio_device/linux/audio_mixer_manager_alsa_linux.cc
@@ -25,7 +25,6 @@ namespace webrtc
 {
 
 AudioMixerManagerLinuxALSA::AudioMixerManagerLinuxALSA(const int32_t id) :
-    _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
     _id(id),
     _outputMixerHandle(NULL),
     _inputMixerHandle(NULL),
@@ -43,10 +42,7 @@ AudioMixerManagerLinuxALSA::~AudioMixerManagerLinuxALSA()
 {
     WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
                  "%s destructed", __FUNCTION__);
-
     Close();
-
-    delete &_critSect;
 }
 
 // ============================================================================
@@ -58,7 +54,7 @@ int32_t AudioMixerManagerLinuxALSA::Close()
     WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s",
                  __FUNCTION__);
 
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
 
     CloseSpeaker();
     CloseMicrophone();
@@ -72,7 +68,7 @@ int32_t AudioMixerManagerLinuxALSA::CloseSpeaker()
     WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s",
                  __FUNCTION__);
 
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
 
     int errVal = 0;
 
@@ -113,7 +109,7 @@ int32_t AudioMixerManagerLinuxALSA::CloseMicrophone()
 {
     WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
 
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
 
     int errVal = 0;
 
@@ -165,7 +161,7 @@ int32_t AudioMixerManagerLinuxALSA::OpenSpeaker(char* deviceName)
     WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
                  "AudioMixerManagerLinuxALSA::OpenSpeaker(name=%s)", deviceName);
 
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
 
     int errVal = 0;
 
@@ -259,7 +255,7 @@ int32_t AudioMixerManagerLinuxALSA::OpenMicrophone(char *deviceName)
                  "AudioMixerManagerLinuxALSA::OpenMicrophone(name=%s)",
                  deviceName);
 
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
 
     int errVal = 0;
 
@@ -378,7 +374,7 @@ int32_t AudioMixerManagerLinuxALSA::SetSpeakerVolume(
                  "AudioMixerManagerLinuxALSA::SetSpeakerVolume(volume=%u)",
                  volume);
 
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
 
     if (_outputMixerElement == NULL)
     {
@@ -642,7 +638,7 @@ int32_t AudioMixerManagerLinuxALSA::SetSpeakerMute(bool enable)
                  "AudioMixerManagerLinuxALSA::SetSpeakerMute(enable=%u)",
                  enable);
 
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
 
     if (_outputMixerElement == NULL)
     {
@@ -739,7 +735,7 @@ int32_t AudioMixerManagerLinuxALSA::SetMicrophoneMute(bool enable)
                  "AudioMixerManagerLinuxALSA::SetMicrophoneMute(enable=%u)",
                  enable);
 
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
 
     if (_inputMixerElement == NULL)
     {
@@ -838,7 +834,7 @@ int32_t AudioMixerManagerLinuxALSA::SetMicrophoneBoost(bool enable)
                  "AudioMixerManagerLinuxALSA::SetMicrophoneBoost(enable=%u)",
                  enable);
 
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
 
     if (_inputMixerHandle == NULL)
     {
@@ -900,7 +896,7 @@ int32_t AudioMixerManagerLinuxALSA::SetMicrophoneVolume(
                  "AudioMixerManagerLinuxALSA::SetMicrophoneVolume(volume=%u)",
                  volume);
 
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
 
     if (_inputMixerElement == NULL)
     {
@@ -1296,19 +1292,16 @@ void AudioMixerManagerLinuxALSA::GetControlName(char* controlName,
     // controlName: "hw:CARD=Intel"
     char* pos1 = strchr(deviceName, ':');
     char* pos2 = strchr(deviceName, ',');
-    if (!pos2)
-    {
+    if (!pos2) {
         // Can also be default:CARD=Intel
         pos2 = &deviceName[strlen(deviceName)];
     }
-    if (pos1 && pos2)
-    {
+    if (pos1 && pos2) {
         strcpy(controlName, "hw");
         int nChar = (int) (pos2 - pos1);
         strncpy(&controlName[2], pos1, nChar);
         controlName[2 + nChar] = '\0';
-    } else
-    {
+    } else {
         strcpy(controlName, deviceName);
     }
 
diff --git a/webrtc/modules/audio_device/linux/audio_mixer_manager_alsa_linux.h b/webrtc/modules/audio_device/linux/audio_mixer_manager_alsa_linux.h
index b8be8c1b70..0ad11c8b6d 100644
--- a/webrtc/modules/audio_device/linux/audio_mixer_manager_alsa_linux.h
+++ b/webrtc/modules/audio_device/linux/audio_mixer_manager_alsa_linux.h
@@ -11,9 +11,9 @@
 #ifndef WEBRTC_AUDIO_DEVICE_AUDIO_MIXER_MANAGER_ALSA_LINUX_H
 #define WEBRTC_AUDIO_DEVICE_AUDIO_MIXER_MANAGER_ALSA_LINUX_H
 
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
 #include "webrtc/modules/audio_device/linux/alsasymboltable_linux.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/typedefs.h"
 
 #include <alsa/asoundlib.h>
@@ -63,7 +63,7 @@ private:
     void GetControlName(char *controlName, char* deviceName) const;
 
 private:
-    CriticalSectionWrapper& _critSect;
+    rtc::CriticalSection _critSect;
     int32_t _id;
     mutable snd_mixer_t* _outputMixerHandle;
     char _outputMixerStr[kAdmMaxDeviceNameSize];
diff --git a/webrtc/modules/audio_device/win/audio_device_core_win.cc b/webrtc/modules/audio_device/win/audio_device_core_win.cc
index 8b38b94f62..9d1d5322fb 100644
--- a/webrtc/modules/audio_device/win/audio_device_core_win.cc
+++ b/webrtc/modules/audio_device/win/audio_device_core_win.cc
@@ -406,8 +406,6 @@ bool AudioDeviceWindowsCore::CoreAudioIsSupported()
 
 AudioDeviceWindowsCore::AudioDeviceWindowsCore(const int32_t id) :
     _comInit(ScopedCOMInitializer::kMTA),
-    _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
-    _volumeMutex(*CriticalSectionWrapper::CreateCriticalSection()),
     _id(id),
     _ptrAudioBuffer(NULL),
     _ptrEnumerator(NULL),
@@ -638,9 +636,6 @@ AudioDeviceWindowsCore::~AudioDeviceWindowsCore()
                 "AudioDeviceWindowsCore::~AudioDeviceWindowsCore() the Avrt DLL module is now unloaded");
         }
     }
-
-    delete &_critSect;
-    delete &_volumeMutex;
 }
 
 // ============================================================================
@@ -680,7 +675,7 @@ int32_t AudioDeviceWindowsCore::ActiveAudioLayer(AudioDeviceModule::AudioLayer&
 // ----------------------------------------------------------------------------
 
 AudioDeviceGeneric::InitStatus AudioDeviceWindowsCore::Init() {
-  CriticalSectionScoped lock(&_critSect);
+  rtc::CritScope lock(&_critSect);
 
   if (_initialized) {
     return InitStatus::OK;
@@ -709,7 +704,7 @@ AudioDeviceGeneric::InitStatus AudioDeviceWindowsCore::Init() {
 int32_t AudioDeviceWindowsCore::Terminate()
 {
 
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
 
     if (!_initialized) {
         return 0;
@@ -751,7 +746,7 @@ bool AudioDeviceWindowsCore::Initialized() const
 int32_t AudioDeviceWindowsCore::InitSpeaker()
 {
 
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
 
     if (_playing)
     {
@@ -833,7 +828,7 @@ int32_t AudioDeviceWindowsCore::InitSpeaker()
 int32_t AudioDeviceWindowsCore::InitMicrophone()
 {
 
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
 
     if (_recording)
     {
@@ -923,7 +918,7 @@ bool AudioDeviceWindowsCore::MicrophoneIsInitialized() const
 int32_t AudioDeviceWindowsCore::SpeakerVolumeIsAvailable(bool& available)
 {
 
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
 
     if (_ptrDeviceOut == NULL)
     {
@@ -968,7 +963,7 @@ int32_t AudioDeviceWindowsCore::SetSpeakerVolume(uint32_t volume)
 {
 
     {
-        CriticalSectionScoped lock(&_critSect);
+        rtc::CritScope lock(&_critSect);
 
         if (!_speakerIsInitialized)
         {
@@ -1011,7 +1006,7 @@ int32_t AudioDeviceWindowsCore::SpeakerVolume(uint32_t& volume) const
 {
 
     {
-        CriticalSectionScoped lock(&_critSect);
+        rtc::CritScope lock(&_critSect);
 
         if (!_speakerIsInitialized)
         {
@@ -1123,7 +1118,7 @@ int32_t AudioDeviceWindowsCore::SpeakerVolumeStepSize(uint16_t& stepSize) const
 int32_t AudioDeviceWindowsCore::SpeakerMuteIsAvailable(bool& available)
 {
 
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
 
     if (_ptrDeviceOut == NULL)
     {
@@ -1162,7 +1157,7 @@ Exit:
 int32_t AudioDeviceWindowsCore::SetSpeakerMute(bool enable)
 {
 
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
 
     if (!_speakerIsInitialized)
     {
@@ -1242,7 +1237,7 @@ Exit:
 int32_t AudioDeviceWindowsCore::MicrophoneMuteIsAvailable(bool& available)
 {
 
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
 
     if (_ptrDeviceIn == NULL)
     {
@@ -1402,7 +1397,7 @@ int32_t AudioDeviceWindowsCore::StereoRecordingIsAvailable(bool& available)
 int32_t AudioDeviceWindowsCore::SetStereoRecording(bool enable)
 {
 
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
 
     if (enable)
     {
@@ -1453,7 +1448,7 @@ int32_t AudioDeviceWindowsCore::StereoPlayoutIsAvailable(bool& available)
 int32_t AudioDeviceWindowsCore::SetStereoPlayout(bool enable)
 {
 
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
 
     if (enable)
     {
@@ -1492,7 +1487,7 @@ int32_t AudioDeviceWindowsCore::StereoPlayout(bool& enabled) const
 
 int32_t AudioDeviceWindowsCore::SetAGC(bool enable)
 {
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
     _AGC = enable;
     return 0;
 }
@@ -1503,7 +1498,7 @@ int32_t AudioDeviceWindowsCore::SetAGC(bool enable)
 
 bool AudioDeviceWindowsCore::AGC() const
 {
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
     return _AGC;
 }
 
@@ -1514,7 +1509,7 @@ bool AudioDeviceWindowsCore::AGC() const
 int32_t AudioDeviceWindowsCore::MicrophoneVolumeIsAvailable(bool& available)
 {
 
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
 
     if (_ptrDeviceIn == NULL)
     {
@@ -1553,7 +1548,7 @@ int32_t AudioDeviceWindowsCore::SetMicrophoneVolume(uint32_t volume)
     WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id, "AudioDeviceWindowsCore::SetMicrophoneVolume(volume=%u)", volume);
 
     {
-        CriticalSectionScoped lock(&_critSect);
+        rtc::CritScope lock(&_critSect);
 
         if (!_microphoneIsInitialized)
         {
@@ -1594,7 +1589,7 @@ Exit:
 int32_t AudioDeviceWindowsCore::MicrophoneVolume(uint32_t& volume) const
 {
     {
-        CriticalSectionScoped lock(&_critSect);
+        rtc::CritScope lock(&_critSect);
 
         if (!_microphoneIsInitialized)
         {
@@ -1689,7 +1684,7 @@ int32_t AudioDeviceWindowsCore::MicrophoneVolumeStepSize(uint16_t& stepSize) con
 int16_t AudioDeviceWindowsCore::PlayoutDevices()
 {
 
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
 
     if (_RefreshDeviceList(eRender) != -1)
     {
@@ -1720,7 +1715,7 @@ int32_t AudioDeviceWindowsCore::SetPlayoutDevice(uint16_t index)
         return -1;
     }
 
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
 
     HRESULT hr(S_OK);
 
@@ -1775,7 +1770,7 @@ int32_t AudioDeviceWindowsCore::SetPlayoutDevice(AudioDeviceModule::WindowsDevic
         role = eCommunications;
     }
 
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
 
     // Refresh the list of rendering endpoint devices
     _RefreshDeviceList(eRender);
@@ -1845,7 +1840,7 @@ int32_t AudioDeviceWindowsCore::PlayoutDeviceName(
         memset(guid, 0, kAdmMaxGuidSize);
     }
 
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
 
     int32_t ret(-1);
     WCHAR szDeviceName[MAX_PATH];
@@ -1925,7 +1920,7 @@ int32_t AudioDeviceWindowsCore::RecordingDeviceName(
         memset(guid, 0, kAdmMaxGuidSize);
     }
 
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
 
     int32_t ret(-1);
     WCHAR szDeviceName[MAX_PATH];
@@ -1979,7 +1974,7 @@ int32_t AudioDeviceWindowsCore::RecordingDeviceName(
 int16_t AudioDeviceWindowsCore::RecordingDevices()
 {
 
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
 
     if (_RefreshDeviceList(eCapture) != -1)
     {
@@ -2010,7 +2005,7 @@ int32_t AudioDeviceWindowsCore::SetRecordingDevice(uint16_t index)
         return -1;
     }
 
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
 
     HRESULT hr(S_OK);
 
@@ -2065,7 +2060,7 @@ int32_t AudioDeviceWindowsCore::SetRecordingDevice(AudioDeviceModule::WindowsDev
         role = eCommunications;
     }
 
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
 
     // Refresh the list of capture endpoint devices
     _RefreshDeviceList(eCapture);
@@ -2155,7 +2150,7 @@ int32_t AudioDeviceWindowsCore::RecordingIsAvailable(bool& available)
 int32_t AudioDeviceWindowsCore::InitPlayout()
 {
 
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
 
     if (_playing)
     {
@@ -2495,7 +2490,7 @@ int32_t AudioDeviceWindowsCore::InitRecordingDMO()
 int32_t AudioDeviceWindowsCore::InitRecording()
 {
 
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
 
     if (_recording)
     {
@@ -2746,7 +2741,7 @@ int32_t AudioDeviceWindowsCore::StartRecording()
     }
 
     {
-        CriticalSectionScoped critScoped(&_critSect);
+        rtc::CritScope critScoped(&_critSect);
 
         // Create thread which will drive the capturing
         LPTHREAD_START_ROUTINE lpStartAddress = WSAPICaptureThread;
@@ -2996,7 +2991,7 @@ int32_t AudioDeviceWindowsCore::StartPlayout()
     }
 
     {
-        CriticalSectionScoped critScoped(&_critSect);
+        rtc::CritScope critScoped(&_critSect);
 
         // Create thread which will drive the rendering.
         assert(_hPlayThread == NULL);
@@ -3046,7 +3041,7 @@ int32_t AudioDeviceWindowsCore::StopPlayout()
     }
 
     {
-        CriticalSectionScoped critScoped(&_critSect) ;
+        rtc::CritScope critScoped(&_critSect) ;
 
         if (_hPlayThread == NULL)
         {
@@ -3079,7 +3074,7 @@ int32_t AudioDeviceWindowsCore::StopPlayout()
     }
 
     {
-        CriticalSectionScoped critScoped(&_critSect);
+        rtc::CritScope critScoped(&_critSect);
         WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
             "webrtc_core_audio_render_thread is now closed");
 
@@ -3123,7 +3118,7 @@ int32_t AudioDeviceWindowsCore::StopPlayout()
 
 int32_t AudioDeviceWindowsCore::PlayoutDelay(uint16_t& delayMS) const
 {
-    CriticalSectionScoped critScoped(&_critSect);
+    rtc::CritScope critScoped(&_critSect);
     delayMS = static_cast<uint16_t>(_sndCardPlayDelay);
     return 0;
 }
@@ -3134,7 +3129,7 @@ int32_t AudioDeviceWindowsCore::PlayoutDelay(uint16_t& delayMS) const
 
 int32_t AudioDeviceWindowsCore::RecordingDelay(uint16_t& delayMS) const
 {
-    CriticalSectionScoped critScoped(&_critSect);
+    rtc::CritScope critScoped(&_critSect);
     delayMS = static_cast<uint16_t>(_sndCardRecDelay);
     return 0;
 }
@@ -3154,7 +3149,7 @@ bool AudioDeviceWindowsCore::Playing() const
 int32_t AudioDeviceWindowsCore::SetPlayoutBuffer(const AudioDeviceModule::BufferType type, uint16_t sizeMS)
 {
 
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
 
     _playBufType = type;
 
@@ -3172,7 +3167,7 @@ int32_t AudioDeviceWindowsCore::SetPlayoutBuffer(const AudioDeviceModule::Buffer
 
 int32_t AudioDeviceWindowsCore::PlayoutBuffer(AudioDeviceModule::BufferType& type, uint16_t& sizeMS) const
 {
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
     type = _playBufType;
 
     if (type == AudioDeviceModule::kFixedBufferSize)
@@ -3773,7 +3768,7 @@ DWORD AudioDeviceWindowsCore::DoCaptureThreadPollDMO()
 
         while (keepRecording)
         {
-            CriticalSectionScoped critScoped(&_critSect);
+            rtc::CritScope critScoped(&_critSect);
 
             DWORD dwStatus = 0;
             {
diff --git a/webrtc/modules/audio_device/win/audio_device_core_win.h b/webrtc/modules/audio_device/win/audio_device_core_win.h
index 5e813cc4a7..0c9ada52ea 100644
--- a/webrtc/modules/audio_device/win/audio_device_core_win.h
+++ b/webrtc/modules/audio_device/win/audio_device_core_win.h
@@ -24,8 +24,8 @@
 #include <endpointvolume.h>
 #include <mediaobj.h>        // IMediaObject
 
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/base/scoped_ref_ptr.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 // Use Multimedia Class Scheduler Service (MMCSS) to boost the thread priority
 #pragma comment( lib, "avrt.lib" )
@@ -237,10 +237,8 @@ private:    // thread functions
     void _Lock() { _critSect.Enter(); };
     void _UnLock() { _critSect.Leave(); };
 
-private:
     int32_t Id() {return _id;}
 
-private:
     int SetDMOProperties();
 
     int SetBoolProperty(IPropertyStore* ptrPS,
@@ -272,21 +270,18 @@ private:
 
     int32_t InitRecordingDMO();
 
-private:
     ScopedCOMInitializer                    _comInit;
     AudioDeviceBuffer*                      _ptrAudioBuffer;
-    CriticalSectionWrapper&                 _critSect;
-    CriticalSectionWrapper&                 _volumeMutex;
-    int32_t                           _id;
+    rtc::CriticalSection                    _critSect;
+    rtc::CriticalSection                    _volumeMutex;
+    int32_t                                 _id;
 
-private:  // MMDevice
     IMMDeviceEnumerator*                    _ptrEnumerator;
     IMMDeviceCollection*                    _ptrRenderCollection;
     IMMDeviceCollection*                    _ptrCaptureCollection;
     IMMDevice*                              _ptrDeviceOut;
     IMMDevice*                              _ptrDeviceIn;
 
-private:  // WASAPI
     IAudioClient*                           _ptrClientOut;
     IAudioClient*                           _ptrClientIn;
     IAudioRenderClient*                     _ptrRenderClient;
diff --git a/webrtc/modules/bitrate_controller/include/bitrate_controller.h b/webrtc/modules/bitrate_controller/include/bitrate_controller.h
index 36637a3dc0..194c99be55 100644
--- a/webrtc/modules/bitrate_controller/include/bitrate_controller.h
+++ b/webrtc/modules/bitrate_controller/include/bitrate_controller.h
@@ -24,7 +24,6 @@
 
 namespace webrtc {
 
-class CriticalSectionWrapper;
 class RtcEventLog;
 
 // Deprecated
diff --git a/webrtc/modules/media_file/media_file_impl.cc b/webrtc/modules/media_file/media_file_impl.cc
index 9f9511d837..e35626c760 100644
--- a/webrtc/modules/media_file/media_file_impl.cc
+++ b/webrtc/modules/media_file/media_file_impl.cc
@@ -12,7 +12,6 @@
 
 #include "webrtc/base/format_macros.h"
 #include "webrtc/modules/media_file/media_file_impl.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/file_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
 
@@ -29,8 +28,6 @@ void MediaFile::DestroyMediaFile(MediaFile* module)
 
 MediaFileImpl::MediaFileImpl(const int32_t id)
     : _id(id),
-      _crit(CriticalSectionWrapper::CreateCriticalSection()),
-      _callbackCrit(CriticalSectionWrapper::CreateCriticalSection()),
       _ptrFileUtilityObj(NULL),
       codec_info_(),
       _ptrInStream(NULL),
@@ -57,7 +54,7 @@ MediaFileImpl::~MediaFileImpl()
 {
     WEBRTC_TRACE(kTraceMemory, kTraceFile, _id, "~MediaFileImpl()");
     {
-        CriticalSectionScoped lock(_crit);
+        rtc::CritScope lock(&_crit);
 
         if(_playingActive)
         {
@@ -79,9 +76,6 @@ MediaFileImpl::~MediaFileImpl()
             _ptrOutStream = NULL;
         }
     }
-
-    delete _crit;
-    delete _callbackCrit;
 }
 
 int64_t MediaFileImpl::TimeUntilNextProcess()
@@ -119,7 +113,7 @@ int32_t MediaFileImpl::PlayoutAudioData(int8_t* buffer,
 
     int32_t bytesRead = 0;
     {
-        CriticalSectionScoped lock(_crit);
+        rtc::CritScope lock(&_crit);
 
         if(!_playingActive)
         {
@@ -213,7 +207,7 @@ void MediaFileImpl::HandlePlayCallbacks(int32_t bytesRead)
     }
 
     // Only _callbackCrit may and should be taken when making callbacks.
-    CriticalSectionScoped lock(_callbackCrit);
+    rtc::CritScope lock(&_callbackCrit);
     if(_ptrCallback)
     {
         if(callbackNotifyMs)
@@ -252,7 +246,7 @@ int32_t MediaFileImpl::PlayoutStereoData(
     bool playEnded = false;
     uint32_t callbackNotifyMs = 0;
     {
-        CriticalSectionScoped lock(_crit);
+        rtc::CritScope lock(&_crit);
 
         if(!_playingActive || !_isStereo)
         {
@@ -313,7 +307,7 @@ int32_t MediaFileImpl::PlayoutStereoData(
         }
     }
 
-    CriticalSectionScoped lock(_callbackCrit);
+    rtc::CritScope lock(&_callbackCrit);
     if(_ptrCallback)
     {
         if(callbackNotifyMs)
@@ -386,7 +380,7 @@ int32_t MediaFileImpl::StartPlayingAudioFile(
         return -1;
     }
 
-    CriticalSectionScoped lock(_crit);
+    rtc::CritScope lock(&_crit);
     _openFile = true;
     strncpy(_fileName, fileName, sizeof(_fileName));
     _fileName[sizeof(_fileName) - 1] = '\0';
@@ -424,7 +418,7 @@ int32_t MediaFileImpl::StartPlayingStream(
         return -1;
     }
 
-    CriticalSectionScoped lock(_crit);
+    rtc::CritScope lock(&_crit);
     if(_playingActive || _recordingActive)
     {
         WEBRTC_TRACE(
@@ -555,7 +549,7 @@ int32_t MediaFileImpl::StartPlayingStream(
 int32_t MediaFileImpl::StopPlaying()
 {
 
-    CriticalSectionScoped lock(_crit);
+    rtc::CritScope lock(&_crit);
     _isStereo = false;
     if(_ptrFileUtilityObj)
     {
@@ -590,7 +584,7 @@ int32_t MediaFileImpl::StopPlaying()
 bool MediaFileImpl::IsPlaying()
 {
     WEBRTC_TRACE(kTraceStream, kTraceFile, _id, "MediaFileImpl::IsPlaying()");
-    CriticalSectionScoped lock(_crit);
+    rtc::CritScope lock(&_crit);
     return _playingActive;
 }
 
@@ -612,7 +606,7 @@ int32_t MediaFileImpl::IncomingAudioData(
     bool recordingEnded = false;
     uint32_t callbackNotifyMs = 0;
     {
-        CriticalSectionScoped lock(_crit);
+        rtc::CritScope lock(&_crit);
 
         if(!_recordingActive)
         {
@@ -707,7 +701,7 @@ int32_t MediaFileImpl::IncomingAudioData(
     }
 
     // Only _callbackCrit may and should be taken when making callbacks.
-    CriticalSectionScoped lock(_callbackCrit);
+    rtc::CritScope lock(&_callbackCrit);
     if(_ptrCallback)
     {
         if(callbackNotifyMs)
@@ -767,7 +761,7 @@ int32_t MediaFileImpl::StartRecordingAudioFile(
         return -1;
     }
 
-    CriticalSectionScoped lock(_crit);
+    rtc::CritScope lock(&_crit);
     _openFile = true;
     strncpy(_fileName, fileName, sizeof(_fileName));
     _fileName[sizeof(_fileName) - 1] = '\0';
@@ -786,7 +780,7 @@ int32_t MediaFileImpl::StartRecordingAudioStream(
         return -1;
     }
 
-    CriticalSectionScoped lock(_crit);
+    rtc::CritScope lock(&_crit);
     if(_recordingActive || _playingActive)
     {
         WEBRTC_TRACE(
@@ -923,7 +917,7 @@ int32_t MediaFileImpl::StartRecordingAudioStream(
 int32_t MediaFileImpl::StopRecording()
 {
 
-    CriticalSectionScoped lock(_crit);
+    rtc::CritScope lock(&_crit);
     if(!_recordingActive)
     {
         WEBRTC_TRACE(kTraceWarning, kTraceFile, _id,
@@ -967,14 +961,14 @@ int32_t MediaFileImpl::StopRecording()
 bool MediaFileImpl::IsRecording()
 {
     WEBRTC_TRACE(kTraceStream, kTraceFile, _id, "MediaFileImpl::IsRecording()");
-    CriticalSectionScoped lock(_crit);
+    rtc::CritScope lock(&_crit);
     return _recordingActive;
 }
 
 int32_t MediaFileImpl::RecordDurationMs(uint32_t& durationMs)
 {
 
-    CriticalSectionScoped lock(_crit);
+    rtc::CritScope lock(&_crit);
     if(!_recordingActive)
     {
         durationMs = 0;
@@ -987,14 +981,14 @@ int32_t MediaFileImpl::RecordDurationMs(uint32_t& durationMs)
 bool MediaFileImpl::IsStereo()
 {
     WEBRTC_TRACE(kTraceStream, kTraceFile, _id, "MediaFileImpl::IsStereo()");
-    CriticalSectionScoped lock(_crit);
+    rtc::CritScope lock(&_crit);
     return _isStereo;
 }
 
 int32_t MediaFileImpl::SetModuleFileCallback(FileCallback* callback)
 {
 
-    CriticalSectionScoped lock(_callbackCrit);
+    rtc::CritScope lock(&_callbackCrit);
 
     _ptrCallback = callback;
     return 0;
@@ -1038,7 +1032,7 @@ int32_t MediaFileImpl::FileDurationMs(const char* fileName,
 
 int32_t MediaFileImpl::PlayoutPositionMs(uint32_t& positionMs) const
 {
-    CriticalSectionScoped lock(_crit);
+    rtc::CritScope lock(&_crit);
     if(!_playingActive)
     {
         positionMs = 0;
@@ -1050,7 +1044,7 @@ int32_t MediaFileImpl::PlayoutPositionMs(uint32_t& positionMs) const
 
 int32_t MediaFileImpl::codec_info(CodecInst& codecInst) const
 {
-    CriticalSectionScoped lock(_crit);
+    rtc::CritScope lock(&_crit);
     if(!_playingActive && !_recordingActive)
     {
         WEBRTC_TRACE(kTraceError, kTraceFile, _id,
diff --git a/webrtc/modules/media_file/media_file_impl.h b/webrtc/modules/media_file/media_file_impl.h
index 677098e8fc..7d974ef313 100644
--- a/webrtc/modules/media_file/media_file_impl.h
+++ b/webrtc/modules/media_file/media_file_impl.h
@@ -11,12 +11,12 @@
 #ifndef WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_IMPL_H_
 #define WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_IMPL_H_
 
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/media_file/media_file.h"
 #include "webrtc/modules/media_file/media_file_defines.h"
 #include "webrtc/modules/media_file/media_file_utility.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
 class MediaFileImpl : public MediaFile
@@ -120,8 +120,8 @@ private:
         const uint32_t stopPointMs);
 
     int32_t _id;
-    CriticalSectionWrapper* _crit;
-    CriticalSectionWrapper* _callbackCrit;
+    rtc::CriticalSection _crit;
+    rtc::CriticalSection _callbackCrit;
 
     ModuleFileUtility* _ptrFileUtilityObj;
     CodecInst codec_info_;
diff --git a/webrtc/modules/pacing/paced_sender.cc b/webrtc/modules/pacing/paced_sender.cc
index 96a6a8860e..a57ec05149 100644
--- a/webrtc/modules/pacing/paced_sender.cc
+++ b/webrtc/modules/pacing/paced_sender.cc
@@ -23,7 +23,6 @@
 #include "webrtc/modules/pacing/bitrate_prober.h"
 #include "webrtc/modules/utility/include/process_thread.h"
 #include "webrtc/system_wrappers/include/clock.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/field_trial.h"
 
 namespace {
@@ -253,7 +252,6 @@ PacedSender::PacedSender(const Clock* clock,
     : clock_(clock),
       packet_sender_(packet_sender),
       alr_detector_(new AlrDetector()),
-      critsect_(CriticalSectionWrapper::CreateCriticalSection()),
       paused_(false),
       media_budget_(new paced_sender::IntervalBudget(0)),
       padding_budget_(new paced_sender::IntervalBudget(0)),
@@ -272,14 +270,14 @@ PacedSender::PacedSender(const Clock* clock,
 PacedSender::~PacedSender() {}
 
 void PacedSender::CreateProbeCluster(int bitrate_bps) {
-  CriticalSectionScoped cs(critsect_.get());
+  rtc::CritScope cs(&critsect_);
   prober_->CreateProbeCluster(bitrate_bps, clock_->TimeInMilliseconds());
 }
 
 void PacedSender::Pause() {
   LOG(LS_INFO) << "PacedSender paused.";
   {
-    CriticalSectionScoped cs(critsect_.get());
+    rtc::CritScope cs(&critsect_);
     paused_ = true;
   }
   // Tell the process thread to call our TimeUntilNextProcess() method to get
@@ -291,7 +289,7 @@ void PacedSender::Pause() {
 void PacedSender::Resume() {
   LOG(LS_INFO) << "PacedSender resumed.";
   {
-    CriticalSectionScoped cs(critsect_.get());
+    rtc::CritScope cs(&critsect_);
     paused_ = false;
   }
   // Tell the process thread to call our TimeUntilNextProcess() method to
@@ -302,14 +300,14 @@ void PacedSender::Resume() {
 
 void PacedSender::SetProbingEnabled(bool enabled) {
   RTC_CHECK_EQ(0, packet_counter_);
-  CriticalSectionScoped cs(critsect_.get());
+  rtc::CritScope cs(&critsect_);
   prober_->SetEnabled(enabled);
 }
 
 void PacedSender::SetEstimatedBitrate(uint32_t bitrate_bps) {
   if (bitrate_bps == 0)
     LOG(LS_ERROR) << "PacedSender is not designed to handle 0 bitrate.";
-  CriticalSectionScoped cs(critsect_.get());
+  rtc::CritScope cs(&critsect_);
   estimated_bitrate_bps_ = bitrate_bps;
   padding_budget_->set_target_rate_kbps(
       std::min(estimated_bitrate_bps_ / 1000, max_padding_bitrate_kbps_));
@@ -321,7 +319,7 @@ void PacedSender::SetEstimatedBitrate(uint32_t bitrate_bps) {
 
 void PacedSender::SetSendBitrateLimits(int min_send_bitrate_bps,
                                        int padding_bitrate) {
-  CriticalSectionScoped cs(critsect_.get());
+  rtc::CritScope cs(&critsect_);
   min_send_bitrate_kbps_ = min_send_bitrate_bps / 1000;
   pacing_bitrate_kbps_ =
       std::max(min_send_bitrate_kbps_, estimated_bitrate_bps_ / 1000) *
@@ -337,7 +335,7 @@ void PacedSender::InsertPacket(RtpPacketSender::Priority priority,
                                int64_t capture_time_ms,
                                size_t bytes,
                                bool retransmission) {
-  CriticalSectionScoped cs(critsect_.get());
+  rtc::CritScope cs(&critsect_);
   RTC_DCHECK(estimated_bitrate_bps_ > 0)
         << "SetEstimatedBitrate must be called before InsertPacket.";
 
@@ -353,7 +351,7 @@ void PacedSender::InsertPacket(RtpPacketSender::Priority priority,
 }
 
 int64_t PacedSender::ExpectedQueueTimeMs() const {
-  CriticalSectionScoped cs(critsect_.get());
+  rtc::CritScope cs(&critsect_);
   RTC_DCHECK_GT(pacing_bitrate_kbps_, 0);
   return static_cast<int64_t>(packets_->SizeInBytes() * 8 /
                               pacing_bitrate_kbps_);
@@ -361,17 +359,17 @@ int64_t PacedSender::ExpectedQueueTimeMs() const {
 
 rtc::Optional<int64_t> PacedSender::GetApplicationLimitedRegionStartTime()
     const {
-  CriticalSectionScoped cs(critsect_.get());
+  rtc::CritScope cs(&critsect_);
   return alr_detector_->GetApplicationLimitedRegionStartTime();
 }
 
 size_t PacedSender::QueueSizePackets() const {
-  CriticalSectionScoped cs(critsect_.get());
+  rtc::CritScope cs(&critsect_);
   return packets_->SizeInPackets();
 }
 
 int64_t PacedSender::QueueInMs() const {
-  CriticalSectionScoped cs(critsect_.get());
+  rtc::CritScope cs(&critsect_);
 
   int64_t oldest_packet = packets_->OldestEnqueueTimeMs();
   if (oldest_packet == 0)
@@ -381,13 +379,13 @@ int64_t PacedSender::QueueInMs() const {
 }
 
 int64_t PacedSender::AverageQueueTimeMs() {
-  CriticalSectionScoped cs(critsect_.get());
+  rtc::CritScope cs(&critsect_);
   packets_->UpdateQueueTime(clock_->TimeInMilliseconds());
   return packets_->AverageQueueTimeMs();
 }
 
 int64_t PacedSender::TimeUntilNextProcess() {
-  CriticalSectionScoped cs(critsect_.get());
+  rtc::CritScope cs(&critsect_);
   if (paused_)
     return 1000 * 60 * 60;
 
@@ -403,7 +401,7 @@ int64_t PacedSender::TimeUntilNextProcess() {
 
 void PacedSender::Process() {
   int64_t now_us = clock_->TimeInMicroseconds();
-  CriticalSectionScoped cs(critsect_.get());
+  rtc::CritScope cs(&critsect_);
   int64_t elapsed_time_ms = (now_us - time_last_update_us_ + 500) / 1000;
   time_last_update_us_ = now_us;
   int target_bitrate_kbps = pacing_bitrate_kbps_;
@@ -489,11 +487,11 @@ bool PacedSender::SendPacket(const paced_sender::Packet& packet,
     return false;
   }
 
-  critsect_->Leave();
+  critsect_.Leave();
   const bool success = packet_sender_->TimeToSendPacket(
       packet.ssrc, packet.sequence_number, packet.capture_time_ms,
       packet.retransmission, pacing_info);
-  critsect_->Enter();
+  critsect_.Enter();
 
   if (success) {
     // TODO(holmer): High priority packets should only be accounted for if we
@@ -509,10 +507,10 @@ bool PacedSender::SendPacket(const paced_sender::Packet& packet,
 
 size_t PacedSender::SendPadding(size_t padding_needed,
                                 const PacedPacketInfo& pacing_info) {
-  critsect_->Leave();
+  critsect_.Leave();
   size_t bytes_sent =
       packet_sender_->TimeToSendPadding(padding_needed, pacing_info);
-  critsect_->Enter();
+  critsect_.Enter();
 
   if (bytes_sent > 0) {
     UpdateBudgetWithBytesSent(bytes_sent);
diff --git a/webrtc/modules/pacing/paced_sender.h b/webrtc/modules/pacing/paced_sender.h
index 320e62f076..3b5fe2e45e 100644
--- a/webrtc/modules/pacing/paced_sender.h
+++ b/webrtc/modules/pacing/paced_sender.h
@@ -15,6 +15,7 @@
 #include <memory>
 #include <set>
 
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/base/optional.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/modules/include/module.h"
@@ -25,7 +26,6 @@ namespace webrtc {
 class AlrDetector;
 class BitrateProber;
 class Clock;
-class CriticalSectionWrapper;
 class ProbeClusterCreatedObserver;
 class RtcEventLog;
 
@@ -162,7 +162,7 @@ class PacedSender : public Module, public RtpPacketSender {
   PacketSender* const packet_sender_;
   std::unique_ptr<AlrDetector> alr_detector_ GUARDED_BY(critsect_);
 
-  std::unique_ptr<CriticalSectionWrapper> critsect_;
+  rtc::CriticalSection critsect_;
   bool paused_ GUARDED_BY(critsect_);
   // This is the media budget, keeping track of how many bits of media
   // we can pace out during the current interval.