diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index 7b9a803..25f44c1 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -14,6 +14,7 @@
 #include <utility>
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/base/format_macros.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/thread_checker.h"
@@ -30,7 +31,6 @@
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
 #include "webrtc/modules/utility/include/audio_frame_operations.h"
 #include "webrtc/modules/utility/include/process_thread.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/include/voe_external_media.h"
@@ -157,9 +157,7 @@
 // Statistics callback, called at each generation of a new RTCP report block.
 class StatisticsProxy : public RtcpStatisticsCallback {
  public:
-  StatisticsProxy(uint32_t ssrc)
-   : stats_lock_(CriticalSectionWrapper::CreateCriticalSection()),
-     ssrc_(ssrc) {}
+  StatisticsProxy(uint32_t ssrc) : ssrc_(ssrc) {}
   virtual ~StatisticsProxy() {}
 
   void StatisticsUpdated(const RtcpStatistics& statistics,
@@ -167,7 +165,7 @@
     if (ssrc != ssrc_)
       return;
 
-    CriticalSectionScoped cs(stats_lock_.get());
+    rtc::CritScope cs(&stats_lock_);
     stats_.rtcp = statistics;
     if (statistics.jitter > stats_.max_jitter) {
       stats_.max_jitter = statistics.jitter;
@@ -177,7 +175,7 @@
   void CNameChanged(const char* cname, uint32_t ssrc) override {}
 
   ChannelStatistics GetStats() {
-    CriticalSectionScoped cs(stats_lock_.get());
+    rtc::CritScope cs(&stats_lock_);
     return stats_;
   }
 
@@ -185,7 +183,7 @@
   // StatisticsUpdated calls are triggered from threads in the RTP module,
   // while GetStats calls can be triggered from the public voice engine API,
   // hence synchronization is needed.
-  rtc::scoped_ptr<CriticalSectionWrapper> stats_lock_;
+  rtc::CriticalSection stats_lock_;
   const uint32_t ssrc_;
   ChannelStatistics stats_;
 };
@@ -298,7 +296,7 @@
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::InFrameType(frame_type=%d)", frame_type);
 
-    CriticalSectionScoped cs(&_callbackCritSect);
+    rtc::CritScope cs(&_callbackCritSect);
     _sendFrameType = (frame_type == kAudioFrameSpeech);
     return 0;
 }
@@ -306,7 +304,7 @@
 int32_t
 Channel::OnRxVadDetected(int vadDecision)
 {
-    CriticalSectionScoped cs(&_callbackCritSect);
+    rtc::CritScope cs(&_callbackCritSect);
     if (_rxVadObserverPtr)
     {
         _rxVadObserverPtr->OnRxVad(_channelId, vadDecision);
@@ -321,7 +319,7 @@
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::SendPacket(channel=%d, len=%" PRIuS ")", len);
 
-    CriticalSectionScoped cs(&_callbackCritSect);
+    rtc::CritScope cs(&_callbackCritSect);
 
     if (_transportPtr == NULL)
     {
@@ -352,7 +350,7 @@
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::SendRtcp(len=%" PRIuS ")", len);
 
-    CriticalSectionScoped cs(&_callbackCritSect);
+    rtc::CritScope cs(&_callbackCritSect);
     if (_transportPtr == NULL)
     {
         WEBRTC_TRACE(kTraceError, kTraceVoice,
@@ -566,7 +564,7 @@
       // scaling/panning, as that applies to the mix operation.
       // External recipients of the audio (e.g. via AudioTrack), will do their
       // own mixing/dynamic processing.
-      CriticalSectionScoped cs(&_callbackCritSect);
+      rtc::CritScope cs(&_callbackCritSect);
       if (audio_sink_) {
         AudioSinkInterface::Data data(
             &audioFrame->data_[0],
@@ -580,7 +578,7 @@
     float left_pan =  1.0f;
     float right_pan =  1.0f;
     {
-      CriticalSectionScoped cs(&volume_settings_critsect_);
+      rtc::CritScope cs(&volume_settings_critsect_);
       output_gain = _outputGain;
       left_pan = _panLeft;
       right_pan= _panRight;
@@ -620,7 +618,7 @@
     // External media
     if (_outputExternalMedia)
     {
-        CriticalSectionScoped cs(&_callbackCritSect);
+        rtc::CritScope cs(&_callbackCritSect);
         const bool isStereo = (audioFrame->num_channels_ == 2);
         if (_outputExternalMediaCallbackPtr)
         {
@@ -633,7 +631,7 @@
 
     // Record playout if enabled
     {
-        CriticalSectionScoped cs(&_fileCritSect);
+        rtc::CritScope cs(&_fileCritSect);
 
         if (_outputFileRecording && _outputFileRecorderPtr)
         {
@@ -660,7 +658,7 @@
           (GetPlayoutFrequency() / 1000);
 
       {
-        CriticalSectionScoped lock(ts_stats_lock_.get());
+        rtc::CritScope lock(&ts_stats_lock_);
         // Compute ntp time.
         audioFrame->ntp_time_ms_ = ntp_estimator_.Estimate(
             audioFrame->timestamp_);
@@ -704,7 +702,7 @@
     // limit the spectrum anyway.
     if (channel_state_.Get().output_file_playing)
     {
-        CriticalSectionScoped cs(&_fileCritSect);
+        rtc::CritScope cs(&_fileCritSect);
         if (_outputFilePlayerPtr)
         {
             if(_outputFilePlayerPtr->Frequency()>highestNeeded)
@@ -790,7 +788,7 @@
 
     assert(id == _outputFileRecorderId);
 
-    CriticalSectionScoped cs(&_fileCritSect);
+    rtc::CritScope cs(&_fileCritSect);
 
     _outputFileRecording = false;
     WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
@@ -803,11 +801,7 @@
                  uint32_t instanceId,
                  RtcEventLog* const event_log,
                  const Config& config)
-    : _fileCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
-      _callbackCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
-      volume_settings_critsect_(
-          *CriticalSectionWrapper::CreateCriticalSection()),
-      _instanceId(instanceId),
+    : _instanceId(instanceId),
       _channelId(channelId),
       event_log_(event_log),
       rtp_header_parser_(RtpHeaderParser::Create()),
@@ -848,7 +842,6 @@
       playout_delay_ms_(0),
       _numberOfDiscardedPackets(0),
       send_sequence_number_(0),
-      ts_stats_lock_(CriticalSectionWrapper::CreateCriticalSection()),
       rtp_ts_wraparound_handler_(new rtc::TimestampWrapAroundHandler()),
       capture_start_rtp_time_stamp_(-1),
       capture_start_ntp_time_ms_(-1),
@@ -875,7 +868,6 @@
       _lastPayloadType(0),
       _includeAudioLevelIndication(false),
       _outputSpeechType(AudioFrame::kNormalSpeech),
-      video_sync_lock_(CriticalSectionWrapper::CreateCriticalSection()),
       _average_jitter_buffer_delay_us(0),
       _previousTimestamp(0),
       _recPacketDelayMs(20),
@@ -885,7 +877,6 @@
       restored_packet_in_use_(false),
       rtcp_observer_(new VoERtcpObserver(this)),
       network_predictor_(new NetworkPredictor(Clock::GetRealTimeClock())),
-      assoc_send_channel_lock_(CriticalSectionWrapper::CreateCriticalSection()),
       associate_send_channel_(ChannelOwner(nullptr)),
       pacing_enabled_(config.Get<VoicePacing>().enabled),
       feedback_observer_proxy_(pacing_enabled_ ? new TransportFeedbackProxy()
@@ -953,7 +944,7 @@
     StopPlayout();
 
     {
-        CriticalSectionScoped cs(&_fileCritSect);
+        rtc::CritScope cs(&_fileCritSect);
         if (_inputFilePlayerPtr)
         {
             _inputFilePlayerPtr->RegisterModuleFileCallback(NULL);
@@ -999,11 +990,6 @@
     _moduleProcessThreadPtr->DeRegisterModule(_rtpRtcpModule.get());
 
     // End of modules shutdown
-
-    // Delete other objects
-    delete &_callbackCritSect;
-    delete &_fileCritSect;
-    delete &volume_settings_critsect_;
 }
 
 int32_t
@@ -1164,7 +1150,7 @@
                               ProcessThread& moduleProcessThread,
                               AudioDeviceModule& audioDeviceModule,
                               VoiceEngineObserver* voiceEngineObserver,
-                              CriticalSectionWrapper* callbackCritSect)
+                              rtc::CriticalSection* callbackCritSect)
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::SetEngineInformation()");
@@ -1187,7 +1173,7 @@
 }
 
 void Channel::SetSink(rtc::scoped_ptr<AudioSinkInterface> sink) {
-  CriticalSectionScoped cs(&_callbackCritSect);
+  rtc::CritScope cs(&_callbackCritSect);
   audio_sink_ = std::move(sink);
 }
 
@@ -1267,7 +1253,7 @@
         _engineStatisticsPtr->SetLastError(
             VE_RTP_RTCP_MODULE_ERROR, kTraceError,
             "StartSend() RTP/RTCP failed to start sending");
-        CriticalSectionScoped cs(&_callbackCritSect);
+        rtc::CritScope cs(&_callbackCritSect);
         channel_state_.SetSending(false);
         return -1;
     }
@@ -1339,7 +1325,7 @@
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::RegisterVoiceEngineObserver()");
-    CriticalSectionScoped cs(&_callbackCritSect);
+    rtc::CritScope cs(&_callbackCritSect);
 
     if (_voiceEngineObserverPtr)
     {
@@ -1357,7 +1343,7 @@
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::DeRegisterVoiceEngineObserver()");
-    CriticalSectionScoped cs(&_callbackCritSect);
+    rtc::CritScope cs(&_callbackCritSect);
 
     if (!_voiceEngineObserverPtr)
     {
@@ -1664,7 +1650,7 @@
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::RegisterExternalTransport()");
 
-    CriticalSectionScoped cs(&_callbackCritSect);
+    rtc::CritScope cs(&_callbackCritSect);
 
     if (_externalTransport)
     {
@@ -1684,7 +1670,7 @@
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::DeRegisterExternalTransport()");
 
-    CriticalSectionScoped cs(&_callbackCritSect);
+    rtc::CritScope cs(&_callbackCritSect);
 
     if (!_transportPtr)
     {
@@ -1828,7 +1814,7 @@
   }
 
   {
-    CriticalSectionScoped lock(ts_stats_lock_.get());
+    rtc::CritScope lock(&ts_stats_lock_);
     ntp_estimator_.UpdateRtcpTimestamp(rtt, ntp_secs, ntp_frac, rtp_timestamp);
   }
   return 0;
@@ -1857,7 +1843,7 @@
     }
 
     {
-        CriticalSectionScoped cs(&_fileCritSect);
+        rtc::CritScope cs(&_fileCritSect);
 
         if (_outputFilePlayerPtr)
         {
@@ -1936,7 +1922,7 @@
     }
 
     {
-      CriticalSectionScoped cs(&_fileCritSect);
+      rtc::CritScope cs(&_fileCritSect);
 
       // Destroy the old instance
       if (_outputFilePlayerPtr)
@@ -1995,7 +1981,7 @@
     }
 
     {
-        CriticalSectionScoped cs(&_fileCritSect);
+        rtc::CritScope cs(&_fileCritSect);
 
         if (_outputFilePlayerPtr->StopPlayingFile() != 0)
         {
@@ -2047,7 +2033,7 @@
     if (_outputMixerPtr->SetAnonymousMixabilityStatus(*this, true) != 0)
     {
         channel_state_.SetOutputFilePlaying(false);
-        CriticalSectionScoped cs(&_fileCritSect);
+        rtc::CritScope cs(&_fileCritSect);
         _engineStatisticsPtr->SetLastError(
             VE_AUDIO_CONF_MIX_MODULE_ERROR, kTraceError,
             "StartPlayingFile() failed to add participant as file to mixer");
@@ -2074,7 +2060,7 @@
                  "stopPosition=%d)", fileName, loop, format, volumeScaling,
                  startPosition, stopPosition);
 
-    CriticalSectionScoped cs(&_fileCritSect);
+    rtc::CritScope cs(&_fileCritSect);
 
     if (channel_state_.Get().input_file_playing)
     {
@@ -2149,7 +2135,7 @@
         return -1;
     }
 
-    CriticalSectionScoped cs(&_fileCritSect);
+    rtc::CritScope cs(&_fileCritSect);
 
     if (channel_state_.Get().input_file_playing)
     {
@@ -2205,7 +2191,7 @@
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::StopPlayingFileAsMicrophone()");
 
-    CriticalSectionScoped cs(&_fileCritSect);
+    rtc::CritScope cs(&_fileCritSect);
 
     if (!channel_state_.Get().input_file_playing)
     {
@@ -2273,7 +2259,7 @@
         format = kFileFormatCompressedFile;
     }
 
-    CriticalSectionScoped cs(&_fileCritSect);
+    rtc::CritScope cs(&_fileCritSect);
 
     // Destroy the old instance
     if (_outputFileRecorderPtr)
@@ -2350,7 +2336,7 @@
         format = kFileFormatCompressedFile;
     }
 
-    CriticalSectionScoped cs(&_fileCritSect);
+    rtc::CritScope cs(&_fileCritSect);
 
     // Destroy the old instance
     if (_outputFileRecorderPtr)
@@ -2401,7 +2387,7 @@
     }
 
 
-    CriticalSectionScoped cs(&_fileCritSect);
+    rtc::CritScope cs(&_fileCritSect);
 
     if (_outputFileRecorderPtr->StopRecording() != 0)
     {
@@ -2421,7 +2407,7 @@
 void
 Channel::SetMixWithMicStatus(bool mix)
 {
-    CriticalSectionScoped cs(&_fileCritSect);
+    rtc::CritScope cs(&_fileCritSect);
     _mixFileWithMicrophone=mix;
 }
 
@@ -2444,7 +2430,7 @@
 int
 Channel::SetMute(bool enable)
 {
-    CriticalSectionScoped cs(&volume_settings_critsect_);
+    rtc::CritScope cs(&volume_settings_critsect_);
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                "Channel::SetMute(enable=%d)", enable);
     _mute = enable;
@@ -2454,14 +2440,14 @@
 bool
 Channel::Mute() const
 {
-    CriticalSectionScoped cs(&volume_settings_critsect_);
+    rtc::CritScope cs(&volume_settings_critsect_);
     return _mute;
 }
 
 int
 Channel::SetOutputVolumePan(float left, float right)
 {
-    CriticalSectionScoped cs(&volume_settings_critsect_);
+    rtc::CritScope cs(&volume_settings_critsect_);
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                "Channel::SetOutputVolumePan()");
     _panLeft = left;
@@ -2472,7 +2458,7 @@
 int
 Channel::GetOutputVolumePan(float& left, float& right) const
 {
-    CriticalSectionScoped cs(&volume_settings_critsect_);
+    rtc::CritScope cs(&volume_settings_critsect_);
     left = _panLeft;
     right = _panRight;
     return 0;
@@ -2481,7 +2467,7 @@
 int
 Channel::SetChannelOutputVolumeScaling(float scaling)
 {
-    CriticalSectionScoped cs(&volume_settings_critsect_);
+    rtc::CritScope cs(&volume_settings_critsect_);
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                "Channel::SetChannelOutputVolumeScaling()");
     _outputGain = scaling;
@@ -2491,7 +2477,7 @@
 int
 Channel::GetChannelOutputVolumeScaling(float& scaling) const
 {
-    CriticalSectionScoped cs(&volume_settings_critsect_);
+    rtc::CritScope cs(&volume_settings_critsect_);
     scaling = _outputGain;
     return 0;
 }
@@ -2601,7 +2587,7 @@
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::RegisterRxVadObserver()");
-    CriticalSectionScoped cs(&_callbackCritSect);
+    rtc::CritScope cs(&_callbackCritSect);
 
     if (_rxVadObserverPtr)
     {
@@ -2620,7 +2606,7 @@
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::DeRegisterRxVadObserver()");
-    CriticalSectionScoped cs(&_callbackCritSect);
+    rtc::CritScope cs(&_callbackCritSect);
 
     if (!_rxVadObserverPtr)
     {
@@ -3260,7 +3246,7 @@
 
     // --- Timestamps
     {
-      CriticalSectionScoped lock(ts_stats_lock_.get());
+      rtc::CritScope lock(&ts_stats_lock_);
       stats.capture_start_ntp_time_ms_ = capture_start_ntp_time_ms_;
     }
     return 0;
@@ -3401,7 +3387,7 @@
 
     if (channel_state_.Get().input_external_media)
     {
-        CriticalSectionScoped cs(&_callbackCritSect);
+        rtc::CritScope cs(&_callbackCritSect);
         const bool isStereo = (_audioFrame.num_channels_ == 2);
         if (_inputExternalMediaCallbackPtr)
         {
@@ -3465,7 +3451,7 @@
 }
 
 void Channel::DisassociateSendChannel(int channel_id) {
-  CriticalSectionScoped lock(assoc_send_channel_lock_.get());
+  rtc::CritScope lock(&assoc_send_channel_lock_);
   Channel* channel = associate_send_channel_.channel();
   if (channel && channel->ChannelId() == channel_id) {
     // If this channel is associated with a send channel of the specified
@@ -3482,7 +3468,7 @@
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::RegisterExternalMediaProcessing()");
 
-    CriticalSectionScoped cs(&_callbackCritSect);
+    rtc::CritScope cs(&_callbackCritSect);
 
     if (kPlaybackPerChannel == type)
     {
@@ -3518,7 +3504,7 @@
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::DeRegisterExternalMediaProcessing()");
 
-    CriticalSectionScoped cs(&_callbackCritSect);
+    rtc::CritScope cs(&_callbackCritSect);
 
     if (kPlaybackPerChannel == type)
     {
@@ -3580,7 +3566,7 @@
 
 bool Channel::GetDelayEstimate(int* jitter_buffer_delay_ms,
                                int* playout_buffer_delay_ms) const {
-  CriticalSectionScoped cs(video_sync_lock_.get());
+  rtc::CritScope lock(&video_sync_lock_);
   if (_average_jitter_buffer_delay_us == 0) {
     return false;
   }
@@ -3627,7 +3613,7 @@
 int Channel::GetPlayoutTimestamp(unsigned int& timestamp) {
   uint32_t playout_timestamp_rtp = 0;
   {
-    CriticalSectionScoped cs(video_sync_lock_.get());
+    rtc::CritScope lock(&video_sync_lock_);
     playout_timestamp_rtp = playout_timestamp_rtp_;
   }
   if (playout_timestamp_rtp == 0)  {
@@ -3681,7 +3667,7 @@
     size_t fileSamples(0);
 
     {
-        CriticalSectionScoped cs(&_fileCritSect);
+        rtc::CritScope cs(&_fileCritSect);
 
         if (_inputFilePlayerPtr == NULL)
         {
@@ -3751,7 +3737,7 @@
     size_t fileSamples(0);
 
     {
-        CriticalSectionScoped cs(&_fileCritSect);
+        rtc::CritScope cs(&_fileCritSect);
 
         if (_outputFilePlayerPtr == NULL)
         {
@@ -3900,7 +3886,7 @@
                playout_timestamp);
 
   {
-    CriticalSectionScoped cs(video_sync_lock_.get());
+    rtc::CritScope lock(&video_sync_lock_);
     if (rtcp) {
       playout_timestamp_rtcp_ = playout_timestamp;
     } else {
@@ -3941,7 +3927,7 @@
   if (timestamp_diff_ms == 0) return;
 
   {
-    CriticalSectionScoped cs(video_sync_lock_.get());
+    rtc::CritScope lock(&video_sync_lock_);
 
     if (packet_delay_ms >= 10 && packet_delay_ms <= 60) {
       _recPacketDelayMs = packet_delay_ms;
@@ -4085,7 +4071,7 @@
   int64_t rtt = 0;
   if (report_blocks.empty()) {
     if (allow_associate_channel) {
-      CriticalSectionScoped lock(assoc_send_channel_lock_.get());
+      rtc::CritScope lock(&assoc_send_channel_lock_);
       Channel* channel = associate_send_channel_.channel();
       // Tries to get RTT from an associated channel. This is important for
       // receive-only channels.
diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h
index 3ef5b4a..3a4383f 100644
--- a/webrtc/voice_engine/channel.h
+++ b/webrtc/voice_engine/channel.h
@@ -47,7 +47,6 @@
 
 class AudioDeviceModule;
 class Config;
-class CriticalSectionWrapper;
 class FileWrapper;
 class PacketRouter;
 class ProcessThread;
@@ -103,57 +102,56 @@
         bool receiving;
     };
 
-    ChannelState() : lock_(CriticalSectionWrapper::CreateCriticalSection()) {
-    }
+    ChannelState() {}
     virtual ~ChannelState() {}
 
     void Reset() {
-        CriticalSectionScoped lock(lock_.get());
+        rtc::CritScope lock(&lock_);
         state_ = State();
     }
 
     State Get() const {
-        CriticalSectionScoped lock(lock_.get());
+        rtc::CritScope lock(&lock_);
         return state_;
     }
 
     void SetRxApmIsEnabled(bool enable) {
-        CriticalSectionScoped lock(lock_.get());
+        rtc::CritScope lock(&lock_);
         state_.rx_apm_is_enabled = enable;
     }
 
     void SetInputExternalMedia(bool enable) {
-        CriticalSectionScoped lock(lock_.get());
+        rtc::CritScope lock(&lock_);
         state_.input_external_media = enable;
     }
 
     void SetOutputFilePlaying(bool enable) {
-        CriticalSectionScoped lock(lock_.get());
+        rtc::CritScope lock(&lock_);
         state_.output_file_playing = enable;
     }
 
     void SetInputFilePlaying(bool enable) {
-        CriticalSectionScoped lock(lock_.get());
+        rtc::CritScope lock(&lock_);
         state_.input_file_playing = enable;
     }
 
     void SetPlaying(bool enable) {
-        CriticalSectionScoped lock(lock_.get());
+        rtc::CritScope lock(&lock_);
         state_.playing = enable;
     }
 
     void SetSending(bool enable) {
-        CriticalSectionScoped lock(lock_.get());
+        rtc::CritScope lock(&lock_);
         state_.sending = enable;
     }
 
     void SetReceiving(bool enable) {
-        CriticalSectionScoped lock(lock_.get());
+        rtc::CritScope lock(&lock_);
         state_.receiving = enable;
     }
 
 private:
- rtc::scoped_ptr<CriticalSectionWrapper> lock_;
+    mutable rtc::CriticalSection lock_;
     State state_;
 };
 
@@ -190,7 +188,7 @@
         ProcessThread& moduleProcessThread,
         AudioDeviceModule& audioDeviceModule,
         VoiceEngineObserver* voiceEngineObserver,
-        CriticalSectionWrapper* callbackCritSect);
+        rtc::CriticalSection* callbackCritSect);
     int32_t UpdateLocalTimeStamp();
 
     void SetSink(rtc::scoped_ptr<AudioSinkInterface> sink);
@@ -430,7 +428,7 @@
     }
     bool ExternalTransport() const
     {
-        CriticalSectionScoped cs(&_callbackCritSect);
+        rtc::CritScope cs(&_callbackCritSect);
         return _externalTransport;
     }
     bool ExternalMixing() const
@@ -460,7 +458,7 @@
     // Used for obtaining RTT for a receive-only channel.
     void set_associate_send_channel(const ChannelOwner& channel) {
       assert(_channelId != channel.channel()->ChannelId());
-      CriticalSectionScoped lock(assoc_send_channel_lock_.get());
+      rtc::CritScope lock(&assoc_send_channel_lock_);
       associate_send_channel_ = channel;
     }
 
@@ -494,9 +492,9 @@
     int32_t GetPlayoutFrequency();
     int64_t GetRTT(bool allow_associate_channel) const;
 
-    CriticalSectionWrapper& _fileCritSect;
-    CriticalSectionWrapper& _callbackCritSect;
-    CriticalSectionWrapper& volume_settings_critsect_;
+    mutable rtc::CriticalSection _fileCritSect;
+    mutable rtc::CriticalSection _callbackCritSect;
+    mutable rtc::CriticalSection volume_settings_critsect_;
     uint32_t _instanceId;
     int32_t _channelId;
 
@@ -544,7 +542,7 @@
     uint16_t send_sequence_number_;
     uint8_t restored_packet_[kVoiceEngineMaxIpPacketSizeBytes];
 
-    rtc::scoped_ptr<CriticalSectionWrapper> ts_stats_lock_;
+    mutable rtc::CriticalSection ts_stats_lock_;
 
     rtc::scoped_ptr<rtc::TimestampWrapAroundHandler> rtp_ts_wraparound_handler_;
     // The rtp timestamp of the first played out audio frame.
@@ -560,7 +558,7 @@
     ProcessThread* _moduleProcessThreadPtr;
     AudioDeviceModule* _audioDeviceModulePtr;
     VoiceEngineObserver* _voiceEngineObserverPtr; // owned by base
-    CriticalSectionWrapper* _callbackCritSectPtr; // owned by base
+    rtc::CriticalSection* _callbackCritSectPtr; // owned by base
     Transport* _transportPtr; // WebRtc socket or external transport
     RMSLevel rms_level_;
     rtc::scoped_ptr<AudioProcessing> rx_audioproc_;  // far end AudioProcessing
@@ -585,7 +583,7 @@
     // VoENetwork
     AudioFrame::SpeechType _outputSpeechType;
     // VoEVideoSync
-    rtc::scoped_ptr<CriticalSectionWrapper> video_sync_lock_;
+    mutable rtc::CriticalSection video_sync_lock_;
     uint32_t _average_jitter_buffer_delay_us GUARDED_BY(video_sync_lock_);
     uint32_t _previousTimestamp;
     uint16_t _recPacketDelayMs GUARDED_BY(video_sync_lock_);
@@ -598,7 +596,7 @@
     rtc::scoped_ptr<VoERtcpObserver> rtcp_observer_;
     rtc::scoped_ptr<NetworkPredictor> network_predictor_;
     // An associated send channel.
-    rtc::scoped_ptr<CriticalSectionWrapper> assoc_send_channel_lock_;
+    mutable rtc::CriticalSection assoc_send_channel_lock_;
     ChannelOwner associate_send_channel_ GUARDED_BY(assoc_send_channel_lock_);
 
     bool pacing_enabled_;
diff --git a/webrtc/voice_engine/channel_manager.cc b/webrtc/voice_engine/channel_manager.cc
index 8452c8b..eac2e50 100644
--- a/webrtc/voice_engine/channel_manager.cc
+++ b/webrtc/voice_engine/channel_manager.cc
@@ -48,7 +48,6 @@
 ChannelManager::ChannelManager(uint32_t instance_id, const Config& config)
     : instance_id_(instance_id),
       last_channel_id_(-1),
-      lock_(CriticalSectionWrapper::CreateCriticalSection()),
       config_(config),
       event_log_(RtcEventLog::Create()) {}
 
@@ -66,7 +65,7 @@
                          event_log_.get(), config);
   ChannelOwner channel_owner(channel);
 
-  CriticalSectionScoped crit(lock_.get());
+  rtc::CritScope crit(&lock_);
 
   channels_.push_back(channel_owner);
 
@@ -74,7 +73,7 @@
 }
 
 ChannelOwner ChannelManager::GetChannel(int32_t channel_id) {
-  CriticalSectionScoped crit(lock_.get());
+  rtc::CritScope crit(&lock_);
 
   for (size_t i = 0; i < channels_.size(); ++i) {
     if (channels_[i].channel()->ChannelId() == channel_id)
@@ -84,7 +83,7 @@
 }
 
 void ChannelManager::GetAllChannels(std::vector<ChannelOwner>* channels) {
-  CriticalSectionScoped crit(lock_.get());
+  rtc::CritScope crit(&lock_);
 
   *channels = channels_;
 }
@@ -95,7 +94,7 @@
   // Channels while holding a lock, but rather when the method returns.
   ChannelOwner reference(NULL);
   {
-    CriticalSectionScoped crit(lock_.get());
+    rtc::CritScope crit(&lock_);
     std::vector<ChannelOwner>::iterator to_delete = channels_.end();
     for (auto it = channels_.begin(); it != channels_.end(); ++it) {
       Channel* channel = it->channel();
@@ -119,14 +118,14 @@
   // lock, but rather when the method returns.
   std::vector<ChannelOwner> references;
   {
-    CriticalSectionScoped crit(lock_.get());
+    rtc::CritScope crit(&lock_);
     references = channels_;
     channels_.clear();
   }
 }
 
 size_t ChannelManager::NumOfChannels() const {
-  CriticalSectionScoped crit(lock_.get());
+  rtc::CritScope crit(&lock_);
   return channels_.size();
 }
 
diff --git a/webrtc/voice_engine/channel_manager.h b/webrtc/voice_engine/channel_manager.h
index 4c0cede..a46d71d 100644
--- a/webrtc/voice_engine/channel_manager.h
+++ b/webrtc/voice_engine/channel_manager.h
@@ -14,10 +14,10 @@
 #include <vector>
 
 #include "webrtc/base/constructormagic.h"
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/call/rtc_event_log.h"
 #include "webrtc/system_wrappers/include/atomic32.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -123,7 +123,7 @@
 
   Atomic32 last_channel_id_;
 
-  rtc::scoped_ptr<CriticalSectionWrapper> lock_;
+  mutable rtc::CriticalSection lock_;
   std::vector<ChannelOwner> channels_;
 
   const Config& config_;
diff --git a/webrtc/voice_engine/dtmf_inband.cc b/webrtc/voice_engine/dtmf_inband.cc
index 309bd54..a6d0905 100644
--- a/webrtc/voice_engine/dtmf_inband.cc
+++ b/webrtc/voice_engine/dtmf_inband.cc
@@ -12,7 +12,6 @@
 
 #include <assert.h>
 
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
 
 namespace webrtc {
@@ -66,7 +65,6 @@
 
 
 DtmfInband::DtmfInband(int32_t id) :
-    _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
     _id(id),
     _outputFrequencyHz(8000),
     _frameLengthSamples(0),
@@ -84,7 +82,6 @@
 
 DtmfInband::~DtmfInband()
 {
-	delete &_critSect;
 }
 
 int
@@ -109,7 +106,7 @@
     return 0;
 }
 
-void 
+void
 DtmfInband::Init()
 {
     _remainingSamples = 0;
@@ -130,7 +127,7 @@
                     int32_t lengthMs,
                     int32_t attenuationDb)
 {
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
 
     if (attenuationDb > 36 || eventCode > 15)
     {
@@ -159,7 +156,7 @@
 int
 DtmfInband::ResetTone()
 {
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
 
     ReInit();
 
@@ -174,7 +171,7 @@
 DtmfInband::StartTone(uint8_t eventCode,
                       int32_t attenuationDb)
 {
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
 
     if (attenuationDb > 36 || eventCode > 15)
     {
@@ -200,7 +197,7 @@
 int
 DtmfInband::StopTone()
 {
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
 
     if (!_playing)
     {
@@ -213,16 +210,16 @@
 }
 
 // Shall be called between tones
-void 
+void
 DtmfInband::ReInit()
 {
     _reinit = true;
 }
 
-bool 
+bool
 DtmfInband::IsAddingTone()
 {
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
     return (_remainingSamples > 0 || _playing);
 }
 
@@ -230,7 +227,7 @@
 DtmfInband::Get10msTone(int16_t output[320],
                         uint16_t& outputSizeInSamples)
 {
-    CriticalSectionScoped lock(&_critSect);
+    rtc::CritScope lock(&_critSect);
     if (DtmfFix_generate(output,
                          _eventCode,
                          _attenuationDb,
@@ -248,6 +245,7 @@
 void
 DtmfInband::UpdateDelaySinceLastTone()
 {
+    rtc::CritScope lock(&_critSect);
     _delaySinceLastToneMS += kDtmfFrameSizeMs;
     // avoid wraparound
     if (_delaySinceLastToneMS > (1<<30))
@@ -259,6 +257,7 @@
 uint32_t
 DtmfInband::DelaySinceLastTone() const
 {
+    rtc::CritScope lock(&_critSect);
     return _delaySinceLastToneMS;
 }
 
diff --git a/webrtc/voice_engine/dtmf_inband.h b/webrtc/voice_engine/dtmf_inband.h
index eb045c9..c2e2b2b 100644
--- a/webrtc/voice_engine/dtmf_inband.h
+++ b/webrtc/voice_engine/dtmf_inband.h
@@ -13,9 +13,9 @@
 
 #include "webrtc/typedefs.h"
 #include "webrtc/voice_engine/voice_engine_defines.h"
+#include "webrtc/base/criticalsection.h"
 
 namespace webrtc {
-class CriticalSectionWrapper;
 
 class DtmfInband
 {
@@ -67,7 +67,7 @@
                                    int16_t length);
 
 private:
-    CriticalSectionWrapper& _critSect;
+    mutable rtc::CriticalSection _critSect;
     int32_t _id;
     uint16_t _outputFrequencyHz;  // {8000, 16000, 32000}
     int16_t _oldOutputLow[2];     // Data needed for oscillator model
diff --git a/webrtc/voice_engine/dtmf_inband_queue.cc b/webrtc/voice_engine/dtmf_inband_queue.cc
index 8619a73..4ab74cd 100644
--- a/webrtc/voice_engine/dtmf_inband_queue.cc
+++ b/webrtc/voice_engine/dtmf_inband_queue.cc
@@ -15,7 +15,6 @@
 
 DtmfInbandQueue::DtmfInbandQueue(int32_t id):
     _id(id),
-    _DtmfCritsect(*CriticalSectionWrapper::CreateCriticalSection()),
     _nextEmptyIndex(0)
 {
     memset(_DtmfKey,0, sizeof(_DtmfKey));
@@ -25,13 +24,12 @@
 
 DtmfInbandQueue::~DtmfInbandQueue()
 {
-    delete &_DtmfCritsect;
 }
 
 int
 DtmfInbandQueue::AddDtmf(uint8_t key, uint16_t len, uint8_t level)
 {
-    CriticalSectionScoped lock(&_DtmfCritsect);
+    rtc::CritScope lock(&_DtmfCritsect);
 
     if (_nextEmptyIndex >= kDtmfInbandMax)
     {
@@ -50,7 +48,7 @@
 int8_t
 DtmfInbandQueue::NextDtmf(uint16_t* len, uint8_t* level)
 {
-    CriticalSectionScoped lock(&_DtmfCritsect);
+    rtc::CritScope lock(&_DtmfCritsect);
 
     if(!PendingDtmf())
     {
@@ -74,14 +72,14 @@
 bool
 DtmfInbandQueue::PendingDtmf()
 {
-    CriticalSectionScoped lock(&_DtmfCritsect);
+    rtc::CritScope lock(&_DtmfCritsect);
     return _nextEmptyIndex > 0;
 }
 
 void
 DtmfInbandQueue::ResetDtmf()
 {
-    CriticalSectionScoped lock(&_DtmfCritsect);
+    rtc::CritScope lock(&_DtmfCritsect);
     _nextEmptyIndex = 0;
 }
 
diff --git a/webrtc/voice_engine/dtmf_inband_queue.h b/webrtc/voice_engine/dtmf_inband_queue.h
index e3b9ce3..08e8018 100644
--- a/webrtc/voice_engine/dtmf_inband_queue.h
+++ b/webrtc/voice_engine/dtmf_inband_queue.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_VOICE_ENGINE_DTMF_INBAND_QUEUE_H
 #define WEBRTC_VOICE_ENGINE_DTMF_INBAND_QUEUE_H
 
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/typedefs.h"
 #include "webrtc/voice_engine/voice_engine_defines.h"
 
@@ -38,7 +38,7 @@
     enum {kDtmfInbandMax = 20};
 
     int32_t _id;
-    CriticalSectionWrapper& _DtmfCritsect;
+    rtc::CriticalSection _DtmfCritsect;
     uint8_t _nextEmptyIndex;
     uint8_t _DtmfKey[kDtmfInbandMax];
     uint16_t _DtmfLen[kDtmfInbandMax];
diff --git a/webrtc/voice_engine/level_indicator.cc b/webrtc/voice_engine/level_indicator.cc
index 68a837e..f44ea8e 100644
--- a/webrtc/voice_engine/level_indicator.cc
+++ b/webrtc/voice_engine/level_indicator.cc
@@ -10,7 +10,6 @@
 
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/voice_engine/level_indicator.h"
 
 namespace webrtc {
@@ -25,7 +24,6 @@
 
 
 AudioLevel::AudioLevel() :
-    _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
     _absMax(0),
     _count(0),
     _currentLevel(0),
@@ -33,12 +31,11 @@
 }
 
 AudioLevel::~AudioLevel() {
-    delete &_critSect;
 }
 
 void AudioLevel::Clear()
 {
-    CriticalSectionScoped cs(&_critSect);
+    rtc::CritScope cs(&_critSect);
     _absMax = 0;
     _count = 0;
     _currentLevel = 0;
@@ -56,7 +53,7 @@
 
     // Protect member access using a lock since this method is called on a
     // dedicated audio thread in the RecordedDataIsAvailable() callback.
-    CriticalSectionScoped cs(&_critSect);
+    rtc::CritScope cs(&_critSect);
 
     if (absValue > _absMax)
     _absMax = absValue;
@@ -88,13 +85,13 @@
 
 int8_t AudioLevel::Level() const
 {
-    CriticalSectionScoped cs(&_critSect);
+    rtc::CritScope cs(&_critSect);
     return _currentLevel;
 }
 
 int16_t AudioLevel::LevelFullRange() const
 {
-    CriticalSectionScoped cs(&_critSect);
+    rtc::CritScope cs(&_critSect);
     return _currentLevelFullRange;
 }
 
diff --git a/webrtc/voice_engine/level_indicator.h b/webrtc/voice_engine/level_indicator.h
index c9b0b14..0f146ce 100644
--- a/webrtc/voice_engine/level_indicator.h
+++ b/webrtc/voice_engine/level_indicator.h
@@ -11,13 +11,13 @@
 #ifndef WEBRTC_VOICE_ENGINE_LEVEL_INDICATOR_H
 #define WEBRTC_VOICE_ENGINE_LEVEL_INDICATOR_H
 
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/typedefs.h"
 #include "webrtc/voice_engine/voice_engine_defines.h"
 
 namespace webrtc {
 
 class AudioFrame;
-class CriticalSectionWrapper;
 namespace voe {
 
 class AudioLevel
@@ -40,7 +40,7 @@
 private:
     enum { kUpdateFrequency = 10};
 
-    CriticalSectionWrapper& _critSect;
+    mutable rtc::CriticalSection _critSect;
 
     int16_t _absMax;
     int16_t _count;
diff --git a/webrtc/voice_engine/monitor_module.cc b/webrtc/voice_engine/monitor_module.cc
index 5ea7357..2d8f4d4 100644
--- a/webrtc/voice_engine/monitor_module.cc
+++ b/webrtc/voice_engine/monitor_module.cc
@@ -8,7 +8,6 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/voice_engine/monitor_module.h"
 
@@ -18,20 +17,18 @@
 
 MonitorModule::MonitorModule() :
     _observerPtr(NULL),
-    _callbackCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
     _lastProcessTime(TickTime::MillisecondTimestamp())
 {
 }
 
 MonitorModule::~MonitorModule()
 {
-    delete &_callbackCritSect;
 }
 
 int32_t
 MonitorModule::RegisterObserver(MonitorObserver& observer)
 {
-    CriticalSectionScoped lock(&_callbackCritSect);
+    rtc::CritScope lock(&_callbackCritSect);
     if (_observerPtr)
     {
         return -1;
@@ -43,7 +40,7 @@
 int32_t
 MonitorModule::DeRegisterObserver()
 {
-    CriticalSectionScoped lock(&_callbackCritSect);
+    rtc::CritScope lock(&_callbackCritSect);
     if (!_observerPtr)
     {
         return 0;
@@ -64,9 +61,9 @@
 MonitorModule::Process()
 {
     _lastProcessTime = TickTime::MillisecondTimestamp();
+    rtc::CritScope lock(&_callbackCritSect);
     if (_observerPtr)
     {
-        CriticalSectionScoped lock(&_callbackCritSect);
         _observerPtr->OnPeriodicProcess();
     }
     return 0;
diff --git a/webrtc/voice_engine/monitor_module.h b/webrtc/voice_engine/monitor_module.h
index fe915b3..5539915 100644
--- a/webrtc/voice_engine/monitor_module.h
+++ b/webrtc/voice_engine/monitor_module.h
@@ -11,6 +11,8 @@
 #ifndef WEBRTC_VOICE_ENGINE_MONITOR_MODULE_H
 #define WEBRTC_VOICE_ENGINE_MONITOR_MODULE_H
 
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/thread_annotations.h"
 #include "webrtc/modules/include/module.h"
 #include "webrtc/typedefs.h"
 #include "webrtc/voice_engine/voice_engine_defines.h"
@@ -25,8 +27,6 @@
 
 
 namespace webrtc {
-class CriticalSectionWrapper;
-
 namespace voe {
 
 class MonitorModule : public Module
@@ -45,8 +45,8 @@
  int32_t Process() override;
 
 private:
-    MonitorObserver* _observerPtr;
-    CriticalSectionWrapper&	_callbackCritSect;
+    rtc::CriticalSection _callbackCritSect;
+    MonitorObserver* _observerPtr GUARDED_BY(_callbackCritSect);
     int64_t _lastProcessTime;
 };
 
diff --git a/webrtc/voice_engine/output_mixer.cc b/webrtc/voice_engine/output_mixer.cc
index 0dacf35..aa50e0b 100644
--- a/webrtc/voice_engine/output_mixer.cc
+++ b/webrtc/voice_engine/output_mixer.cc
@@ -13,7 +13,6 @@
 #include "webrtc/base/format_macros.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/utility/include/audio_frame_operations.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"
 #include "webrtc/voice_engine/include/voe_external_media.h"
@@ -68,7 +67,7 @@
                  "OutputMixer::RecordFileEnded(id=%d)", id);
     assert(id == _instanceId);
 
-    CriticalSectionScoped cs(&_fileCritSect);
+    rtc::CritScope cs(&_fileCritSect);
     _outputFileRecording = false;
     WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
                  "OutputMixer::RecordFileEnded() =>"
@@ -92,8 +91,6 @@
 }
 
 OutputMixer::OutputMixer(uint32_t instanceId) :
-    _callbackCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
-    _fileCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
     _mixerModule(*AudioConferenceMixer::Create(instanceId)),
     _audioLevel(),
     _dtmfGenerator(instanceId),
@@ -138,7 +135,7 @@
         DeRegisterExternalMediaProcessing();
     }
     {
-        CriticalSectionScoped cs(&_fileCritSect);
+        rtc::CritScope cs(&_fileCritSect);
         if (_outputFileRecorderPtr)
         {
             _outputFileRecorderPtr->RegisterModuleFileCallback(NULL);
@@ -149,8 +146,6 @@
     }
     _mixerModule.UnRegisterMixedStreamCallback();
     delete &_mixerModule;
-    delete &_callbackCritSect;
-    delete &_fileCritSect;
 }
 
 int32_t
@@ -178,7 +173,7 @@
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
                "OutputMixer::RegisterExternalMediaProcessing()");
 
-    CriticalSectionScoped cs(&_callbackCritSect);
+    rtc::CritScope cs(&_callbackCritSect);
     _externalMediaCallbackPtr = &proccess_object;
     _externalMedia = true;
 
@@ -190,7 +185,7 @@
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
                  "OutputMixer::DeRegisterExternalMediaProcessing()");
 
-    CriticalSectionScoped cs(&_callbackCritSect);
+    rtc::CritScope cs(&_callbackCritSect);
     _externalMedia = false;
     _externalMediaCallbackPtr = NULL;
 
@@ -314,7 +309,7 @@
         format = kFileFormatCompressedFile;
     }
 
-    CriticalSectionScoped cs(&_fileCritSect);
+    rtc::CritScope cs(&_fileCritSect);
 
     // Destroy the old instance
     if (_outputFileRecorderPtr)
@@ -394,7 +389,7 @@
         format = kFileFormatCompressedFile;
     }
 
-    CriticalSectionScoped cs(&_fileCritSect);
+    rtc::CritScope cs(&_fileCritSect);
 
     // Destroy the old instance
     if (_outputFileRecorderPtr)
@@ -445,7 +440,7 @@
         return -1;
     }
 
-    CriticalSectionScoped cs(&_fileCritSect);
+    rtc::CritScope cs(&_fileCritSect);
 
     if (_outputFileRecorderPtr->StopRecording() != 0)
     {
@@ -472,7 +467,7 @@
 
   // --- Record playout if enabled
   {
-    CriticalSectionScoped cs(&_fileCritSect);
+    rtc::CritScope cs(&_fileCritSect);
     if (_outputFileRecording && _outputFileRecorderPtr)
       _outputFileRecorderPtr->RecordAudioToFile(_audioFrame);
   }
@@ -536,7 +531,7 @@
 
     // --- External media processing
     {
-        CriticalSectionScoped cs(&_callbackCritSect);
+        rtc::CritScope cs(&_callbackCritSect);
         if (_externalMedia)
         {
             const bool is_stereo = (_audioFrame.num_channels_ == 2);
diff --git a/webrtc/voice_engine/output_mixer.h b/webrtc/voice_engine/output_mixer.h
index 91387e6..3287860 100644
--- a/webrtc/voice_engine/output_mixer.h
+++ b/webrtc/voice_engine/output_mixer.h
@@ -11,6 +11,7 @@
 #ifndef WEBRTC_VOICE_ENGINE_OUTPUT_MIXER_H_
 #define WEBRTC_VOICE_ENGINE_OUTPUT_MIXER_H_
 
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer.h"
@@ -23,7 +24,6 @@
 namespace webrtc {
 
 class AudioProcessing;
-class CriticalSectionWrapper;
 class FileWrapper;
 class VoEMediaProcess;
 
@@ -108,10 +108,9 @@
     Statistics* _engineStatisticsPtr;
     AudioProcessing* _audioProcessingModulePtr;
 
-    // owns
-    CriticalSectionWrapper& _callbackCritSect;
+    rtc::CriticalSection _callbackCritSect;
     // protect the _outputFileRecorderPtr and _outputFileRecording
-    CriticalSectionWrapper& _fileCritSect;
+    rtc::CriticalSection _fileCritSect;
     AudioConferenceMixer& _mixerModule;
     AudioFrame _audioFrame;
     // Converts mixed audio to the audio device output rate.
diff --git a/webrtc/voice_engine/shared_data.cc b/webrtc/voice_engine/shared_data.cc
index 4bc5b3c..1d50c3c 100644
--- a/webrtc/voice_engine/shared_data.cc
+++ b/webrtc/voice_engine/shared_data.cc
@@ -11,7 +11,6 @@
 #include "webrtc/voice_engine/shared_data.h"
 
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/voice_engine/channel.h"
 #include "webrtc/voice_engine/output_mixer.h"
@@ -25,7 +24,6 @@
 
 SharedData::SharedData(const Config& config)
     : _instanceId(++_gInstanceCounter),
-      _apiCritPtr(CriticalSectionWrapper::CreateCriticalSection()),
       _channelManager(_gInstanceCounter, config),
       _engineStatistics(_gInstanceCounter),
       _audioDevicePtr(NULL),
@@ -51,7 +49,6 @@
     if (_audioDevicePtr) {
         _audioDevicePtr->Release();
     }
-    delete _apiCritPtr;
     _moduleProcessThreadPtr->Stop();
     Trace::ReturnTrace();
 }
diff --git a/webrtc/voice_engine/shared_data.h b/webrtc/voice_engine/shared_data.h
index 9c3d4b1..d9551cf 100644
--- a/webrtc/voice_engine/shared_data.h
+++ b/webrtc/voice_engine/shared_data.h
@@ -11,6 +11,7 @@
 #ifndef WEBRTC_VOICE_ENGINE_SHARED_DATA_H
 #define WEBRTC_VOICE_ENGINE_SHARED_DATA_H
 
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
@@ -23,7 +24,6 @@
 
 namespace webrtc {
 class Config;
-class CriticalSectionWrapper;
 
 namespace voe {
 
@@ -43,7 +43,7 @@
     void set_audio_processing(AudioProcessing* audio_processing);
     TransmitMixer* transmit_mixer() { return _transmitMixerPtr; }
     OutputMixer* output_mixer() { return _outputMixerPtr; }
-    CriticalSectionWrapper* crit_sec() { return _apiCritPtr; }
+    rtc::CriticalSection* crit_sec() { return &_apiCritPtr; }
     ProcessThread* process_thread() { return _moduleProcessThreadPtr.get(); }
     AudioDeviceModule::AudioLayer audio_device_layer() const {
       return _audioDeviceLayer;
@@ -63,7 +63,7 @@
 
 protected:
     const uint32_t _instanceId;
-    CriticalSectionWrapper* _apiCritPtr;
+    mutable rtc::CriticalSection _apiCritPtr;
     ChannelManager _channelManager;
     Statistics _engineStatistics;
     AudioDeviceModule* _audioDevicePtr;
diff --git a/webrtc/voice_engine/statistics.cc b/webrtc/voice_engine/statistics.cc
index 3787e67..0d8250f 100644
--- a/webrtc/voice_engine/statistics.cc
+++ b/webrtc/voice_engine/statistics.cc
@@ -13,7 +13,6 @@
 
 #include "webrtc/voice_engine/statistics.h"
 
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
 
 namespace webrtc {
@@ -21,20 +20,14 @@
 namespace voe {
 
 Statistics::Statistics(uint32_t instanceId) :
-    _critPtr(CriticalSectionWrapper::CreateCriticalSection()),
     _instanceId(instanceId),
     _lastError(0),
     _isInitialized(false)
 {
 }
-	
+
 Statistics::~Statistics()
 {
-    if (_critPtr)
-    {
-        delete _critPtr;
-        _critPtr = NULL;
-    }
 }
 
 int32_t Statistics::SetInitialized()
@@ -56,7 +49,7 @@
 
 int32_t Statistics::SetLastError(int32_t error) const
 {
-    CriticalSectionScoped cs(_critPtr);
+    rtc::CritScope cs(&lock_);
     _lastError = error;
     return 0;
 }
@@ -64,11 +57,11 @@
 int32_t Statistics::SetLastError(int32_t error,
                                  TraceLevel level) const
 {
-    CriticalSectionScoped cs(_critPtr);
-    _lastError = error;
     WEBRTC_TRACE(level, kTraceVoice, VoEId(_instanceId,-1),
                  "error code is set to %d",
-                 _lastError);
+                 error);
+    rtc::CritScope cs(&lock_);
+    _lastError = error;
     return 0;
 }
 
@@ -76,22 +69,28 @@
     int32_t error,
     TraceLevel level, const char* msg) const
 {
-    CriticalSectionScoped cs(_critPtr);
     char traceMessage[KTraceMaxMessageSize];
     assert(strlen(msg) < KTraceMaxMessageSize);
-    _lastError = error;
     sprintf(traceMessage, "%s (error=%d)", msg, error);
+
     WEBRTC_TRACE(level, kTraceVoice, VoEId(_instanceId,-1), "%s",
                  traceMessage);
+
+    rtc::CritScope cs(&lock_);
+    _lastError = error;
     return 0;
 }
 
 int32_t Statistics::LastError() const
 {
-    CriticalSectionScoped cs(_critPtr);
-    WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
-               "LastError() => %d", _lastError);
-    return _lastError;
+    int32_t ret;
+    {
+        rtc::CritScope cs(&lock_);
+        ret = _lastError;
+    }
+    WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
+                 "LastError() => %d", ret);
+    return ret;
 }
 
 }  // namespace voe
diff --git a/webrtc/voice_engine/statistics.h b/webrtc/voice_engine/statistics.h
index 3dab916..e853e68 100644
--- a/webrtc/voice_engine/statistics.h
+++ b/webrtc/voice_engine/statistics.h
@@ -11,14 +11,13 @@
 #ifndef WEBRTC_VOICE_ENGINE_STATISTICS_H
 #define WEBRTC_VOICE_ENGINE_STATISTICS_H
 
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/common_types.h"
 #include "webrtc/typedefs.h"
 #include "webrtc/voice_engine/include/voe_errors.h"
 #include "webrtc/voice_engine/voice_engine_defines.h"
 
 namespace webrtc {
-class CriticalSectionWrapper;
-
 namespace voe {
 
 class Statistics
@@ -40,7 +39,7 @@
     int32_t LastError() const;
 
  private:
-    CriticalSectionWrapper* _critPtr;
+    mutable rtc::CriticalSection lock_;
     const uint32_t _instanceId;
     mutable int32_t _lastError;
     bool _isInitialized;
diff --git a/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc b/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc
index 70f6829..086eeab 100644
--- a/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc
+++ b/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc
@@ -37,9 +37,7 @@
 namespace voetest {
 
 ConferenceTransport::ConferenceTransport()
-    : pq_crit_(webrtc::CriticalSectionWrapper::CreateCriticalSection()),
-      stream_crit_(webrtc::CriticalSectionWrapper::CreateCriticalSection()),
-      packet_event_(webrtc::EventWrapper::Create()),
+    : packet_event_(webrtc::EventWrapper::Create()),
       thread_(Run, this, "ConferenceTransport"),
       rtt_ms_(0),
       stream_count_(0),
@@ -120,7 +118,7 @@
 
 int ConferenceTransport::GetReceiverChannelForSsrc(unsigned int sender_ssrc)
     const {
-  webrtc::CriticalSectionScoped lock(stream_crit_.get());
+  rtc::CritScope lock(&stream_crit_);
   auto it = streams_.find(sender_ssrc);
   if (it != streams_.end()) {
     return it->second.second;
@@ -132,7 +130,7 @@
                                       const void* data,
                                       size_t len) {
   {
-    webrtc::CriticalSectionScoped lock(pq_crit_.get());
+    rtc::CritScope lock(&pq_crit_);
     packet_queue_.push_back(Packet(type, data, len, rtc::Time()));
   }
   packet_event_->Set();
@@ -198,7 +196,7 @@
   while (true) {
     Packet packet;
     {
-      webrtc::CriticalSectionScoped lock(pq_crit_.get());
+      rtc::CritScope lock(&pq_crit_);
       if (packet_queue_.empty())
         break;
       packet = packet_queue_.front();
@@ -245,14 +243,14 @@
   EXPECT_EQ(0, local_rtp_rtcp_->SetLocalSSRC(new_receiver, kLocalSsrc));
 
   {
-    webrtc::CriticalSectionScoped lock(stream_crit_.get());
+    rtc::CritScope lock(&stream_crit_);
     streams_[remote_ssrc] = std::make_pair(new_sender, new_receiver);
   }
   return remote_ssrc;  // remote ssrc used as stream id.
 }
 
 bool ConferenceTransport::RemoveStream(unsigned int id) {
-  webrtc::CriticalSectionScoped lock(stream_crit_.get());
+  rtc::CritScope lock(&stream_crit_);
   auto it = streams_.find(id);
   if (it == streams_.end()) {
     return false;
diff --git a/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h b/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h
index 5d105aa..6b780e3 100644
--- a/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h
+++ b/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h
@@ -17,11 +17,11 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/basictypes.h"
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/base/platform_thread.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/include/voe_codec.h"
@@ -128,17 +128,16 @@
   void SendPacket(const Packet& packet);
   bool DispatchPackets();
 
-  const rtc::scoped_ptr<webrtc::CriticalSectionWrapper> pq_crit_;
-  const rtc::scoped_ptr<webrtc::CriticalSectionWrapper> stream_crit_;
+  mutable rtc::CriticalSection pq_crit_;
+  mutable rtc::CriticalSection stream_crit_;
   const rtc::scoped_ptr<webrtc::EventWrapper> packet_event_;
   rtc::PlatformThread thread_;
 
   unsigned int rtt_ms_;
   unsigned int stream_count_;
 
-  std::map<unsigned int, std::pair<int, int>> streams_
-    GUARDED_BY(stream_crit_.get());
-  std::deque<Packet> packet_queue_ GUARDED_BY(pq_crit_.get());
+  std::map<unsigned int, std::pair<int, int>> streams_ GUARDED_BY(stream_crit_);
+  std::deque<Packet> packet_queue_ GUARDED_BY(pq_crit_);
 
   int local_sender_;  // Channel Id of local sender
   int reflector_;
diff --git a/webrtc/voice_engine/test/auto_test/fixtures/after_initialization_fixture.h b/webrtc/voice_engine/test/auto_test/fixtures/after_initialization_fixture.h
index 116ff0a..843be84 100644
--- a/webrtc/voice_engine/test/auto_test/fixtures/after_initialization_fixture.h
+++ b/webrtc/voice_engine/test/auto_test/fixtures/after_initialization_fixture.h
@@ -13,12 +13,12 @@
 
 #include <deque>
 
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/base/platform_thread.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/system_wrappers/include/atomic32.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/system_wrappers/include/sleep.h"
 #include "webrtc/voice_engine/test/auto_test/fixtures/before_initialization_fixture.h"
@@ -28,8 +28,7 @@
 class LoopBackTransport : public webrtc::Transport {
  public:
   LoopBackTransport(webrtc::VoENetwork* voe_network, int channel)
-      : crit_(webrtc::CriticalSectionWrapper::CreateCriticalSection()),
-        packet_event_(webrtc::EventWrapper::Create()),
+      : packet_event_(webrtc::EventWrapper::Create()),
         thread_(NetworkProcess, this, "LoopBackTransport"),
         channel_(channel),
         voe_network_(voe_network),
@@ -62,7 +61,7 @@
   }
 
   void AddChannel(uint32_t ssrc, int channel) {
-    webrtc::CriticalSectionScoped lock(crit_.get());
+    rtc::CritScope lock(&crit_);
     channels_[ssrc] = channel;
   }
 
@@ -85,7 +84,7 @@
                    const void* data,
                    size_t len) {
     {
-      webrtc::CriticalSectionScoped lock(crit_.get());
+      rtc::CritScope lock(&crit_);
       packet_queue_.push_back(Packet(type, data, len));
     }
     packet_event_->Set();
@@ -110,7 +109,7 @@
       Packet p;
       int channel = channel_;
       {
-        webrtc::CriticalSectionScoped lock(crit_.get());
+        rtc::CritScope lock(&crit_);
         if (packet_queue_.empty())
           break;
         p = packet_queue_.front();
@@ -143,12 +142,12 @@
     return true;
   }
 
-  const rtc::scoped_ptr<webrtc::CriticalSectionWrapper> crit_;
+  mutable rtc::CriticalSection crit_;
   const rtc::scoped_ptr<webrtc::EventWrapper> packet_event_;
   rtc::PlatformThread thread_;
-  std::deque<Packet> packet_queue_ GUARDED_BY(crit_.get());
+  std::deque<Packet> packet_queue_ GUARDED_BY(crit_);
   const int channel_;
-  std::map<uint32_t, int> channels_ GUARDED_BY(crit_.get());
+  std::map<uint32_t, int> channels_ GUARDED_BY(crit_);
   webrtc::VoENetwork* const voe_network_;
   webrtc::Atomic32 transmitted_packets_;
 };
diff --git a/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc b/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc
index 6efa55d..d653cc1 100644
--- a/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc
+++ b/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/system_wrappers/include/atomic32.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/voice_engine/test/auto_test/fixtures/after_streaming_fixture.h"
@@ -17,9 +17,7 @@
 
 class TestRtpObserver : public webrtc::VoERTPObserver {
  public:
-  TestRtpObserver()
-      : crit_(voetest::CriticalSectionWrapper::CreateCriticalSection()),
-        changed_ssrc_event_(voetest::EventWrapper::Create()) {}
+  TestRtpObserver() : changed_ssrc_event_(voetest::EventWrapper::Create()) {}
   virtual ~TestRtpObserver() {}
   virtual void OnIncomingCSRCChanged(int channel,
                                      unsigned int CSRC,
@@ -31,11 +29,11 @@
     EXPECT_EQ(voetest::kEventSignaled, changed_ssrc_event_->Wait(10*1000));
   }
   void SetIncomingSsrc(unsigned int ssrc) {
-    voetest::CriticalSectionScoped lock(crit_.get());
+    rtc::CritScope lock(&crit_);
     incoming_ssrc_ = ssrc;
   }
  public:
-  rtc::scoped_ptr<voetest::CriticalSectionWrapper> crit_;
+  rtc::CriticalSection crit_;
   unsigned int incoming_ssrc_;
   rtc::scoped_ptr<voetest::EventWrapper> changed_ssrc_event_;
 };
@@ -48,7 +46,7 @@
   TEST_LOG("%s", msg);
 
   {
-    voetest::CriticalSectionScoped lock(crit_.get());
+    rtc::CritScope lock(&crit_);
     if (incoming_ssrc_ == SSRC)
       changed_ssrc_event_->Set();
   }
diff --git a/webrtc/voice_engine/test/auto_test/voe_standard_test.h b/webrtc/voice_engine/test/auto_test/voe_standard_test.h
index b925959..38688b3 100644
--- a/webrtc/voice_engine/test/auto_test/voe_standard_test.h
+++ b/webrtc/voice_engine/test/auto_test/voe_standard_test.h
@@ -43,7 +43,6 @@
 
 #ifdef WEBRTC_VOICE_ENGINE_NETEQ_STATS_API
 namespace webrtc {
-class CriticalSectionWrapper;
 class VoENetEqStats;
 }
 #endif
diff --git a/webrtc/voice_engine/transmit_mixer.cc b/webrtc/voice_engine/transmit_mixer.cc
index 1204b04..1490350 100644
--- a/webrtc/voice_engine/transmit_mixer.cc
+++ b/webrtc/voice_engine/transmit_mixer.cc
@@ -13,7 +13,6 @@
 #include "webrtc/base/format_macros.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/modules/utility/include/audio_frame_operations.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/voice_engine/channel.h"
@@ -37,7 +36,7 @@
     bool send_typing_noise_warning = false;
     bool typing_noise_detected = false;
     {
-      CriticalSectionScoped cs(&_critSect);
+      rtc::CritScope cs(&_critSect);
       if (_typingNoiseWarningPending) {
         send_typing_noise_warning = true;
         typing_noise_detected = _typingNoiseDetected;
@@ -45,7 +44,7 @@
       }
     }
     if (send_typing_noise_warning) {
-        CriticalSectionScoped cs(&_callbackCritSect);
+        rtc::CritScope cs(&_callbackCritSect);
         if (_voiceEngineObserverPtr) {
             if (typing_noise_detected) {
                 WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
@@ -71,7 +70,7 @@
       // Modify |_saturationWarning| under lock to avoid conflict with write op
       // in ProcessAudio and also ensure that we don't hold the lock during the
       // callback.
-      CriticalSectionScoped cs(&_critSect);
+      rtc::CritScope cs(&_critSect);
       saturationWarning = _saturationWarning;
       if (_saturationWarning)
         _saturationWarning = false;
@@ -79,7 +78,7 @@
 
     if (saturationWarning)
     {
-        CriticalSectionScoped cs(&_callbackCritSect);
+        rtc::CritScope cs(&_callbackCritSect);
         if (_voiceEngineObserverPtr)
         {
             WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
@@ -118,7 +117,7 @@
 
     assert(id == _filePlayerId);
 
-    CriticalSectionScoped cs(&_critSect);
+    rtc::CritScope cs(&_critSect);
 
     _filePlaying = false;
     WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
@@ -134,14 +133,14 @@
 
     if (id == _fileRecorderId)
     {
-        CriticalSectionScoped cs(&_critSect);
+        rtc::CritScope cs(&_critSect);
         _fileRecording = false;
         WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
                      "TransmitMixer::RecordFileEnded() => fileRecorder module"
                      "is shutdown");
     } else if (id == _fileCallRecorderId)
     {
-        CriticalSectionScoped cs(&_critSect);
+        rtc::CritScope cs(&_critSect);
         _fileCallRecording = false;
         WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
                      "TransmitMixer::RecordFileEnded() => fileCallRecorder"
@@ -193,8 +192,6 @@
     _fileRecording(false),
     _fileCallRecording(false),
     _audioLevel(),
-    _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
-    _callbackCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
 #ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
     _typingNoiseWarningPending(false),
     _typingNoiseDetected(false),
@@ -226,7 +223,7 @@
     DeRegisterExternalMediaProcessing(kRecordingAllChannelsMixed);
     DeRegisterExternalMediaProcessing(kRecordingPreprocessing);
     {
-        CriticalSectionScoped cs(&_critSect);
+        rtc::CritScope cs(&_critSect);
         if (_fileRecorderPtr)
         {
             _fileRecorderPtr->RegisterModuleFileCallback(NULL);
@@ -249,8 +246,6 @@
             _filePlayerPtr = NULL;
         }
     }
-    delete &_critSect;
-    delete &_callbackCritSect;
 }
 
 int32_t
@@ -276,7 +271,7 @@
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
                  "TransmitMixer::RegisterVoiceEngineObserver()");
-    CriticalSectionScoped cs(&_callbackCritSect);
+    rtc::CritScope cs(&_callbackCritSect);
 
     if (_voiceEngineObserverPtr)
     {
@@ -340,7 +335,7 @@
                        samplesPerSec);
 
     {
-      CriticalSectionScoped cs(&_callbackCritSect);
+      rtc::CritScope cs(&_callbackCritSect);
       if (external_preproc_ptr_) {
         external_preproc_ptr_->Process(-1, kRecordingPreprocessing,
                                        _audioFrame.data_,
@@ -388,7 +383,7 @@
     // --- Record to file
     bool file_recording = false;
     {
-        CriticalSectionScoped cs(&_critSect);
+        rtc::CritScope cs(&_critSect);
         file_recording =  _fileRecording;
     }
     if (file_recording)
@@ -397,7 +392,7 @@
     }
 
     {
-      CriticalSectionScoped cs(&_callbackCritSect);
+      rtc::CritScope cs(&_callbackCritSect);
       if (external_postproc_ptr_) {
         external_postproc_ptr_->Process(-1, kRecordingAllChannelsMixed,
                                         _audioFrame.data_,
@@ -520,7 +515,7 @@
         return 0;
     }
 
-    CriticalSectionScoped cs(&_critSect);
+    rtc::CritScope cs(&_critSect);
 
     // Destroy the old instance
     if (_filePlayerPtr)
@@ -597,7 +592,7 @@
         return 0;
     }
 
-    CriticalSectionScoped cs(&_critSect);
+    rtc::CritScope cs(&_critSect);
 
     // Destroy the old instance
     if (_filePlayerPtr)
@@ -654,7 +649,7 @@
         return 0;
     }
 
-    CriticalSectionScoped cs(&_critSect);
+    rtc::CritScope cs(&_critSect);
 
     if (_filePlayerPtr->StopPlayingFile() != 0)
     {
@@ -686,7 +681,7 @@
                  "TransmitMixer::StartRecordingMicrophone(fileName=%s)",
                  fileName);
 
-    CriticalSectionScoped cs(&_critSect);
+    rtc::CritScope cs(&_critSect);
 
     if (_fileRecording)
     {
@@ -764,7 +759,7 @@
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
                "TransmitMixer::StartRecordingMicrophone()");
 
-    CriticalSectionScoped cs(&_critSect);
+    rtc::CritScope cs(&_critSect);
 
     if (_fileRecording)
     {
@@ -841,7 +836,7 @@
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
                  "TransmitMixer::StopRecordingMicrophone()");
 
-    CriticalSectionScoped cs(&_critSect);
+    rtc::CritScope cs(&_critSect);
 
     if (!_fileRecording)
     {
@@ -903,7 +898,7 @@
         format = kFileFormatCompressedFile;
     }
 
-    CriticalSectionScoped cs(&_critSect);
+    rtc::CritScope cs(&_critSect);
 
     // Destroy the old instance
     if (_fileCallRecorderPtr)
@@ -981,7 +976,7 @@
         format = kFileFormatCompressedFile;
     }
 
-    CriticalSectionScoped cs(&_critSect);
+    rtc::CritScope cs(&_critSect);
 
     // Destroy the old instance
     if (_fileCallRecorderPtr)
@@ -1032,7 +1027,7 @@
         return -1;
     }
 
-    CriticalSectionScoped cs(&_critSect);
+    rtc::CritScope cs(&_critSect);
 
     if (_fileCallRecorderPtr->StopRecording() != 0)
     {
@@ -1062,7 +1057,7 @@
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
                "TransmitMixer::RegisterExternalMediaProcessing()");
 
-  CriticalSectionScoped cs(&_callbackCritSect);
+  rtc::CritScope cs(&_callbackCritSect);
   if (!object) {
     return -1;
   }
@@ -1082,7 +1077,7 @@
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
                "TransmitMixer::DeRegisterExternalMediaProcessing()");
 
-  CriticalSectionScoped cs(&_callbackCritSect);
+  rtc::CritScope cs(&_callbackCritSect);
   if (type == kRecordingAllChannelsMixed) {
     external_postproc_ptr_ = NULL;
   } else if (type == kRecordingPreprocessing) {
@@ -1127,7 +1122,7 @@
 
 bool TransmitMixer::IsRecordingMic()
 {
-    CriticalSectionScoped cs(&_critSect);
+    rtc::CritScope cs(&_critSect);
     return _fileRecording;
 }
 
@@ -1162,7 +1157,7 @@
 int32_t TransmitMixer::RecordAudioToFile(
     uint32_t mixingFrequency)
 {
-    CriticalSectionScoped cs(&_critSect);
+    rtc::CritScope cs(&_critSect);
     if (_fileRecorderPtr == NULL)
     {
         WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
@@ -1189,7 +1184,7 @@
 
     size_t fileSamples(0);
     {
-        CriticalSectionScoped cs(&_critSect);
+        rtc::CritScope cs(&_critSect);
         if (_filePlayerPtr == NULL)
         {
             WEBRTC_TRACE(kTraceWarning, kTraceVoice,
@@ -1267,7 +1262,7 @@
   // Store new capture level. Only updated when analog AGC is enabled.
   _captureLevel = agc->stream_analog_level();
 
-  CriticalSectionScoped cs(&_critSect);
+  rtc::CritScope cs(&_critSect);
   // Triggers a callback in OnPeriodicProcess().
   _saturationWarning |= agc->stream_is_saturated();
 }
@@ -1282,11 +1277,11 @@
 
   bool vadActive = _audioFrame.vad_activity_ == AudioFrame::kVadActive;
   if (_typingDetection.Process(keyPressed, vadActive)) {
-    CriticalSectionScoped cs(&_critSect);
+    rtc::CritScope cs(&_critSect);
     _typingNoiseWarningPending = true;
     _typingNoiseDetected = true;
   } else {
-    CriticalSectionScoped cs(&_critSect);
+    rtc::CritScope cs(&_critSect);
     // If there is already a warning pending, do not change the state.
     // Otherwise set a warning pending if last callback was for noise detected.
     if (!_typingNoiseWarningPending && _typingNoiseDetected) {
diff --git a/webrtc/voice_engine/transmit_mixer.h b/webrtc/voice_engine/transmit_mixer.h
index 0aee106..e0246cf 100644
--- a/webrtc/voice_engine/transmit_mixer.h
+++ b/webrtc/voice_engine/transmit_mixer.h
@@ -11,6 +11,7 @@
 #ifndef WEBRTC_VOICE_ENGINE_TRANSMIT_MIXER_H
 #define WEBRTC_VOICE_ENGINE_TRANSMIT_MIXER_H
 
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
 #include "webrtc/common_types.h"
@@ -210,8 +211,8 @@
     bool _fileCallRecording;
     voe::AudioLevel _audioLevel;
     // protect file instances and their variables in MixedParticipants()
-    CriticalSectionWrapper& _critSect;
-    CriticalSectionWrapper& _callbackCritSect;
+    rtc::CriticalSection _critSect;
+    rtc::CriticalSection _callbackCritSect;
 
 #ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
     webrtc::TypingDetection _typingDetection;
diff --git a/webrtc/voice_engine/voe_audio_processing_impl.cc b/webrtc/voice_engine/voe_audio_processing_impl.cc
index 4479fe0..7e082fd 100644
--- a/webrtc/voice_engine/voe_audio_processing_impl.cc
+++ b/webrtc/voice_engine/voe_audio_processing_impl.cc
@@ -12,7 +12,6 @@
 
 #include "webrtc/base/logging.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/voice_engine/channel.h"
 #include "webrtc/voice_engine/include/voe_errors.h"
diff --git a/webrtc/voice_engine/voe_base_impl.cc b/webrtc/voice_engine/voe_base_impl.cc
index 3e5cfbb..e3dee0f 100644
--- a/webrtc/voice_engine/voe_base_impl.cc
+++ b/webrtc/voice_engine/voe_base_impl.cc
@@ -17,7 +17,6 @@
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_device/audio_device_impl.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/file_wrapper.h"
 #include "webrtc/voice_engine/channel.h"
 #include "webrtc/voice_engine/include/voe_errors.h"
@@ -39,16 +38,14 @@
 
 VoEBaseImpl::VoEBaseImpl(voe::SharedData* shared)
     : voiceEngineObserverPtr_(nullptr),
-      callbackCritSect_(*CriticalSectionWrapper::CreateCriticalSection()),
       shared_(shared) {}
 
 VoEBaseImpl::~VoEBaseImpl() {
   TerminateInternal();
-  delete &callbackCritSect_;
 }
 
 void VoEBaseImpl::OnErrorIsReported(const ErrorCode error) {
-  CriticalSectionScoped cs(&callbackCritSect_);
+  rtc::CritScope cs(&callbackCritSect_);
   int errCode = 0;
   if (error == AudioDeviceObserver::kRecordingError) {
     errCode = VE_RUNTIME_REC_ERROR;
@@ -64,7 +61,7 @@
 }
 
 void VoEBaseImpl::OnWarningIsReported(const WarningCode warning) {
-  CriticalSectionScoped cs(&callbackCritSect_);
+  rtc::CritScope cs(&callbackCritSect_);
   int warningCode = 0;
   if (warning == AudioDeviceObserver::kRecordingWarning) {
     warningCode = VE_RUNTIME_REC_WARNING;
@@ -176,7 +173,7 @@
 }
 
 int VoEBaseImpl::RegisterVoiceEngineObserver(VoiceEngineObserver& observer) {
-  CriticalSectionScoped cs(&callbackCritSect_);
+  rtc::CritScope cs(&callbackCritSect_);
   if (voiceEngineObserverPtr_) {
     shared_->SetLastError(
         VE_INVALID_OPERATION, kTraceError,
@@ -196,7 +193,7 @@
 }
 
 int VoEBaseImpl::DeRegisterVoiceEngineObserver() {
-  CriticalSectionScoped cs(&callbackCritSect_);
+  rtc::CritScope cs(&callbackCritSect_);
   if (!voiceEngineObserverPtr_) {
     shared_->SetLastError(
         VE_INVALID_OPERATION, kTraceError,
@@ -216,7 +213,7 @@
 
 int VoEBaseImpl::Init(AudioDeviceModule* external_adm,
                       AudioProcessing* audioproc) {
-  CriticalSectionScoped cs(shared_->crit_sec());
+  rtc::CritScope cs(shared_->crit_sec());
   WebRtcSpl_Init();
   if (shared_->statistics().Initialized()) {
     return 0;
@@ -382,12 +379,12 @@
 }
 
 int VoEBaseImpl::Terminate() {
-  CriticalSectionScoped cs(shared_->crit_sec());
+  rtc::CritScope cs(shared_->crit_sec());
   return TerminateInternal();
 }
 
 int VoEBaseImpl::CreateChannel() {
-  CriticalSectionScoped cs(shared_->crit_sec());
+  rtc::CritScope cs(shared_->crit_sec());
   if (!shared_->statistics().Initialized()) {
     shared_->SetLastError(VE_NOT_INITED, kTraceError);
     return -1;
@@ -398,7 +395,7 @@
 }
 
 int VoEBaseImpl::CreateChannel(const Config& config) {
-  CriticalSectionScoped cs(shared_->crit_sec());
+  rtc::CritScope cs(shared_->crit_sec());
   if (!shared_->statistics().Initialized()) {
     shared_->SetLastError(VE_NOT_INITED, kTraceError);
     return -1;
@@ -434,7 +431,7 @@
 }
 
 int VoEBaseImpl::DeleteChannel(int channel) {
-  CriticalSectionScoped cs(shared_->crit_sec());
+  rtc::CritScope cs(shared_->crit_sec());
   if (!shared_->statistics().Initialized()) {
     shared_->SetLastError(VE_NOT_INITED, kTraceError);
     return -1;
@@ -461,7 +458,7 @@
 }
 
 int VoEBaseImpl::StartReceive(int channel) {
-  CriticalSectionScoped cs(shared_->crit_sec());
+  rtc::CritScope cs(shared_->crit_sec());
   if (!shared_->statistics().Initialized()) {
     shared_->SetLastError(VE_NOT_INITED, kTraceError);
     return -1;
@@ -477,7 +474,7 @@
 }
 
 int VoEBaseImpl::StopReceive(int channel) {
-  CriticalSectionScoped cs(shared_->crit_sec());
+  rtc::CritScope cs(shared_->crit_sec());
   if (!shared_->statistics().Initialized()) {
     shared_->SetLastError(VE_NOT_INITED, kTraceError);
     return -1;
@@ -493,7 +490,7 @@
 }
 
 int VoEBaseImpl::StartPlayout(int channel) {
-  CriticalSectionScoped cs(shared_->crit_sec());
+  rtc::CritScope cs(shared_->crit_sec());
   if (!shared_->statistics().Initialized()) {
     shared_->SetLastError(VE_NOT_INITED, kTraceError);
     return -1;
@@ -517,7 +514,7 @@
 }
 
 int VoEBaseImpl::StopPlayout(int channel) {
-  CriticalSectionScoped cs(shared_->crit_sec());
+  rtc::CritScope cs(shared_->crit_sec());
   if (!shared_->statistics().Initialized()) {
     shared_->SetLastError(VE_NOT_INITED, kTraceError);
     return -1;
@@ -537,7 +534,7 @@
 }
 
 int VoEBaseImpl::StartSend(int channel) {
-  CriticalSectionScoped cs(shared_->crit_sec());
+  rtc::CritScope cs(shared_->crit_sec());
   if (!shared_->statistics().Initialized()) {
     shared_->SetLastError(VE_NOT_INITED, kTraceError);
     return -1;
@@ -561,7 +558,7 @@
 }
 
 int VoEBaseImpl::StopSend(int channel) {
-  CriticalSectionScoped cs(shared_->crit_sec());
+  rtc::CritScope cs(shared_->crit_sec());
   if (!shared_->statistics().Initialized()) {
     shared_->SetLastError(VE_NOT_INITED, kTraceError);
     return -1;
@@ -795,7 +792,7 @@
 
 int VoEBaseImpl::AssociateSendChannel(int channel,
                                       int accociate_send_channel) {
-  CriticalSectionScoped cs(shared_->crit_sec());
+  rtc::CritScope cs(shared_->crit_sec());
 
   if (!shared_->statistics().Initialized()) {
       shared_->SetLastError(VE_NOT_INITED, kTraceError);
diff --git a/webrtc/voice_engine/voe_base_impl.h b/webrtc/voice_engine/voe_base_impl.h
index 58e0387..9fc18ad 100644
--- a/webrtc/voice_engine/voe_base_impl.h
+++ b/webrtc/voice_engine/voe_base_impl.h
@@ -13,6 +13,7 @@
 
 #include "webrtc/voice_engine/include/voe_base.h"
 
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/voice_engine/shared_data.h"
 
@@ -138,7 +139,7 @@
   // channel.
   int InitializeChannel(voe::ChannelOwner* channel_owner);
   VoiceEngineObserver* voiceEngineObserverPtr_;
-  CriticalSectionWrapper& callbackCritSect_;
+  rtc::CriticalSection callbackCritSect_;
 
   AudioFrame audioFrame_;
   voe::SharedData* shared_;
diff --git a/webrtc/voice_engine/voe_codec_impl.cc b/webrtc/voice_engine/voe_codec_impl.cc
index 6eb11b7..5a16592 100644
--- a/webrtc/voice_engine/voe_codec_impl.cc
+++ b/webrtc/voice_engine/voe_codec_impl.cc
@@ -12,7 +12,6 @@
 
 #include "webrtc/base/format_macros.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/voice_engine/channel.h"
 #include "webrtc/voice_engine/include/voe_errors.h"
diff --git a/webrtc/voice_engine/voe_dtmf_impl.cc b/webrtc/voice_engine/voe_dtmf_impl.cc
index cf6ab1d..c8cc52c 100644
--- a/webrtc/voice_engine/voe_dtmf_impl.cc
+++ b/webrtc/voice_engine/voe_dtmf_impl.cc
@@ -10,7 +10,7 @@
 
 #include "webrtc/voice_engine/voe_dtmf_impl.h"
 
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/voice_engine/channel.h"
 #include "webrtc/voice_engine/include/voe_errors.h"
@@ -197,7 +197,7 @@
                "SetDtmfFeedbackStatus(enable=%d, directFeeback=%d)",
                (int)enable, (int)directFeedback);
 
-  CriticalSectionScoped sc(_shared->crit_sec());
+  rtc::CritScope cs(_shared->crit_sec());
 
   _dtmfFeedback = enable;
   _dtmfDirectFeedback = directFeedback;
@@ -206,7 +206,7 @@
 }
 
 int VoEDtmfImpl::GetDtmfFeedbackStatus(bool& enabled, bool& directFeedback) {
-  CriticalSectionScoped sc(_shared->crit_sec());
+  rtc::CritScope cs(_shared->crit_sec());
 
   enabled = _dtmfFeedback;
   directFeedback = _dtmfDirectFeedback;
diff --git a/webrtc/voice_engine/voe_external_media_impl.cc b/webrtc/voice_engine/voe_external_media_impl.cc
index ccb12b9..7b2b6a0 100644
--- a/webrtc/voice_engine/voe_external_media_impl.cc
+++ b/webrtc/voice_engine/voe_external_media_impl.cc
@@ -10,7 +10,6 @@
 
 #include "webrtc/voice_engine/voe_external_media_impl.h"
 
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/voice_engine/channel.h"
 #include "webrtc/voice_engine/include/voe_errors.h"
diff --git a/webrtc/voice_engine/voe_file_impl.cc b/webrtc/voice_engine/voe_file_impl.cc
index 2091e70..339fba1 100644
--- a/webrtc/voice_engine/voe_file_impl.cc
+++ b/webrtc/voice_engine/voe_file_impl.cc
@@ -11,7 +11,6 @@
 #include "webrtc/voice_engine/voe_file_impl.h"
 
 #include "webrtc/modules/media_file/media_file.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"
 #include "webrtc/voice_engine/channel.h"
diff --git a/webrtc/voice_engine/voe_hardware_impl.cc b/webrtc/voice_engine/voe_hardware_impl.cc
index 30eb418..d01caa9 100644
--- a/webrtc/voice_engine/voe_hardware_impl.cc
+++ b/webrtc/voice_engine/voe_hardware_impl.cc
@@ -12,7 +12,6 @@
 
 #include <assert.h>
 
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/voice_engine/include/voe_errors.h"
 #include "webrtc/voice_engine/voice_engine_impl.h"
@@ -234,7 +233,7 @@
   WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                "SetRecordingDevice(index=%d, recordingChannel=%d)", index,
                (int)recordingChannel);
-  CriticalSectionScoped cs(_shared->crit_sec());
+  rtc::CritScope cs(_shared->crit_sec());
 
   if (!_shared->statistics().Initialized()) {
     _shared->SetLastError(VE_NOT_INITED, kTraceError);
@@ -345,7 +344,7 @@
 int VoEHardwareImpl::SetPlayoutDevice(int index) {
   WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                "SetPlayoutDevice(index=%d)", index);
-  CriticalSectionScoped cs(_shared->crit_sec());
+  rtc::CritScope cs(_shared->crit_sec());
 
   if (!_shared->statistics().Initialized()) {
     _shared->SetLastError(VE_NOT_INITED, kTraceError);
diff --git a/webrtc/voice_engine/voe_neteq_stats_impl.cc b/webrtc/voice_engine/voe_neteq_stats_impl.cc
index 807325b..6b2b37c 100644
--- a/webrtc/voice_engine/voe_neteq_stats_impl.cc
+++ b/webrtc/voice_engine/voe_neteq_stats_impl.cc
@@ -11,7 +11,6 @@
 #include "webrtc/voice_engine/voe_neteq_stats_impl.h"
 
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/voice_engine/channel.h"
 #include "webrtc/voice_engine/include/voe_errors.h"
diff --git a/webrtc/voice_engine/voe_network_impl.cc b/webrtc/voice_engine/voe_network_impl.cc
index 0574aa9..5562048 100644
--- a/webrtc/voice_engine/voe_network_impl.cc
+++ b/webrtc/voice_engine/voe_network_impl.cc
@@ -13,7 +13,6 @@
 #include "webrtc/base/checks.h"
 #include "webrtc/base/format_macros.h"
 #include "webrtc/base/logging.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/voice_engine/channel.h"
 #include "webrtc/voice_engine/include/voe_errors.h"
diff --git a/webrtc/voice_engine/voe_rtp_rtcp_impl.cc b/webrtc/voice_engine/voe_rtp_rtcp_impl.cc
index 008445c..ab70fee 100644
--- a/webrtc/voice_engine/voe_rtp_rtcp_impl.cc
+++ b/webrtc/voice_engine/voe_rtp_rtcp_impl.cc
@@ -8,7 +8,6 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/file_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/voice_engine/include/voe_errors.h"
diff --git a/webrtc/voice_engine/voe_video_sync_impl.cc b/webrtc/voice_engine/voe_video_sync_impl.cc
index 77517c6..03d69b4 100644
--- a/webrtc/voice_engine/voe_video_sync_impl.cc
+++ b/webrtc/voice_engine/voe_video_sync_impl.cc
@@ -10,7 +10,6 @@
 
 #include "webrtc/voice_engine/voe_video_sync_impl.h"
 
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/voice_engine/channel.h"
 #include "webrtc/voice_engine/include/voe_errors.h"
diff --git a/webrtc/voice_engine/voe_volume_control_impl.cc b/webrtc/voice_engine/voe_volume_control_impl.cc
index bb57ed7..8beba11 100644
--- a/webrtc/voice_engine/voe_volume_control_impl.cc
+++ b/webrtc/voice_engine/voe_volume_control_impl.cc
@@ -10,7 +10,6 @@
 
 #include "webrtc/voice_engine/voe_volume_control_impl.h"
 
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/voice_engine/channel.h"
 #include "webrtc/voice_engine/include/voe_errors.h"
diff --git a/webrtc/voice_engine/voice_engine_impl.cc b/webrtc/voice_engine/voice_engine_impl.cc
index 7caf933..d0b7412 100644
--- a/webrtc/voice_engine/voice_engine_impl.cc
+++ b/webrtc/voice_engine/voice_engine_impl.cc
@@ -17,7 +17,6 @@
 
 #include "webrtc/base/checks.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/voice_engine/channel_proxy.h"
 #include "webrtc/voice_engine/voice_engine_impl.h"
@@ -66,7 +65,7 @@
 rtc::scoped_ptr<voe::ChannelProxy> VoiceEngineImpl::GetChannelProxy(
     int channel_id) {
   RTC_DCHECK(channel_id >= 0);
-  CriticalSectionScoped cs(crit_sec());
+  rtc::CritScope cs(crit_sec());
   RTC_DCHECK(statistics().Initialized());
   return rtc::scoped_ptr<voe::ChannelProxy>(
       new voe::ChannelProxy(channel_manager().GetChannel(channel_id)));