diff --git a/webrtc/modules/audio_coding/acm2/acm_receiver.cc b/webrtc/modules/audio_coding/acm2/acm_receiver.cc
index f45d5d3414..53ec8841e5 100644
--- a/webrtc/modules/audio_coding/acm2/acm_receiver.cc
+++ b/webrtc/modules/audio_coding/acm2/acm_receiver.cc
@@ -25,7 +25,6 @@
 #include "webrtc/modules/audio_coding/acm2/call_statistics.h"
 #include "webrtc/modules/audio_coding/neteq/include/neteq.h"
 #include "webrtc/system_wrappers/include/clock.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/system_wrappers/include/trace.h"
 
@@ -119,8 +118,7 @@ bool IsCng(int codec_id) {
 }  // namespace
 
 AcmReceiver::AcmReceiver(const AudioCodingModule::Config& config)
-    : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
-      id_(config.id),
+    : id_(config.id),
       last_audio_decoder_(nullptr),
       previous_audio_activity_(AudioFrame::kVadPassive),
       audio_buffer_(new int16_t[AudioFrame::kMaxDataSizeSamples]),
@@ -157,7 +155,7 @@ int AcmReceiver::LeastRequiredDelayMs() const {
 }
 
 rtc::Optional<int> AcmReceiver::last_packet_sample_rate_hz() const {
-  CriticalSectionScoped lock(crit_sect_.get());
+  rtc::CritScope lock(&crit_sect_);
   return last_packet_sample_rate_hz_;
 }
 
@@ -171,7 +169,7 @@ int AcmReceiver::InsertPacket(const WebRtcRTPHeader& rtp_header,
   const RTPHeader* header = &rtp_header.header;  // Just a shorthand.
 
   {
-    CriticalSectionScoped lock(crit_sect_.get());
+    rtc::CritScope lock(&crit_sect_);
 
     const Decoder* decoder = RtpHeaderToDecoder(*header, incoming_payload[0]);
     if (!decoder) {
@@ -216,7 +214,7 @@ int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) {
   size_t num_channels;
 
   // Accessing members, take the lock.
-  CriticalSectionScoped lock(crit_sect_.get());
+  rtc::CritScope lock(&crit_sect_);
 
   // Always write the output to |audio_buffer_| first.
   if (neteq_->GetAudio(AudioFrame::kMaxDataSizeSamples,
@@ -317,7 +315,7 @@ int32_t AcmReceiver::AddCodec(int acm_codec_id,
     return *ned;
   }();
 
-  CriticalSectionScoped lock(crit_sect_.get());
+  rtc::CritScope lock(&crit_sect_);
 
   // The corresponding NetEq decoder ID.
   // If this codec has been registered before.
@@ -366,13 +364,13 @@ int32_t AcmReceiver::AddCodec(int acm_codec_id,
 
 void AcmReceiver::EnableVad() {
   neteq_->EnableVad();
-  CriticalSectionScoped lock(crit_sect_.get());
+  rtc::CritScope lock(&crit_sect_);
   vad_enabled_ = true;
 }
 
 void AcmReceiver::DisableVad() {
   neteq_->DisableVad();
-  CriticalSectionScoped lock(crit_sect_.get());
+  rtc::CritScope lock(&crit_sect_);
   vad_enabled_ = false;
 }
 
@@ -384,7 +382,7 @@ void AcmReceiver::FlushBuffers() {
 // many as it can.
 int AcmReceiver::RemoveAllCodecs() {
   int ret_val = 0;
-  CriticalSectionScoped lock(crit_sect_.get());
+  rtc::CritScope lock(&crit_sect_);
   for (auto it = decoders_.begin(); it != decoders_.end(); ) {
     auto cur = it;
     ++it;  // it will be valid even if we erase cur
@@ -404,7 +402,7 @@ int AcmReceiver::RemoveAllCodecs() {
 }
 
 int AcmReceiver::RemoveCodec(uint8_t payload_type) {
-  CriticalSectionScoped lock(crit_sect_.get());
+  rtc::CritScope lock(&crit_sect_);
   auto it = decoders_.find(payload_type);
   if (it == decoders_.end()) {  // Such a payload-type is not registered.
     return 0;
@@ -422,7 +420,7 @@ int AcmReceiver::RemoveCodec(uint8_t payload_type) {
 }
 
 void AcmReceiver::set_id(int id) {
-  CriticalSectionScoped lock(crit_sect_.get());
+  rtc::CritScope lock(&crit_sect_);
   id_ = id;
 }
 
@@ -431,7 +429,7 @@ bool AcmReceiver::GetPlayoutTimestamp(uint32_t* timestamp) {
 }
 
 int AcmReceiver::LastAudioCodec(CodecInst* codec) const {
-  CriticalSectionScoped lock(crit_sect_.get());
+  rtc::CritScope lock(&crit_sect_);
   if (!last_audio_decoder_) {
     return -1;
   }
@@ -468,7 +466,7 @@ void AcmReceiver::GetNetworkStatistics(NetworkStatistics* acm_stat) {
 
 int AcmReceiver::DecoderByPayloadType(uint8_t payload_type,
                                       CodecInst* codec) const {
-  CriticalSectionScoped lock(crit_sect_.get());
+  rtc::CritScope lock(&crit_sect_);
   auto it = decoders_.find(payload_type);
   if (it == decoders_.end()) {
     LOG(LERROR) << "AcmReceiver::DecoderByPayloadType "
@@ -532,7 +530,7 @@ uint32_t AcmReceiver::NowInTimestamp(int decoder_sampling_rate) const {
 
 void AcmReceiver::GetDecodingCallStatistics(
     AudioDecodingCallStats* stats) const {
-  CriticalSectionScoped lock(crit_sect_.get());
+  rtc::CritScope lock(&crit_sect_);
   *stats = call_stats_.GetDecodingStatistics();
 }
 
diff --git a/webrtc/modules/audio_coding/acm2/acm_receiver.h b/webrtc/modules/audio_coding/acm2/acm_receiver.h
index b150612f69..826cb45864 100644
--- a/webrtc/modules/audio_coding/acm2/acm_receiver.h
+++ b/webrtc/modules/audio_coding/acm2/acm_receiver.h
@@ -16,15 +16,16 @@
 #include <vector>
 
 #include "webrtc/base/array_view.h"
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/base/optional.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/common_audio/vad/include/webrtc_vad.h"
 #include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/acm2/acm_resampler.h"
 #include "webrtc/modules/audio_coding/acm2/call_statistics.h"
 #include "webrtc/modules/audio_coding/acm2/initial_delay_manager.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/neteq/include/neteq.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
@@ -32,7 +33,6 @@
 namespace webrtc {
 
 struct CodecInst;
-class CriticalSectionWrapper;
 class NetEq;
 
 namespace acm2 {
@@ -281,7 +281,7 @@ class AcmReceiver {
 
   uint32_t NowInTimestamp(int decoder_sampling_rate) const;
 
-  rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
+  mutable rtc::CriticalSection crit_sect_;
   int id_;  // TODO(henrik.lundin) Make const.
   const Decoder* last_audio_decoder_ GUARDED_BY(crit_sect_);
   AudioFrame::VADActivity previous_audio_activity_ GUARDED_BY(crit_sect_);
diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc b/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc
index ac302f0fe3..d0e02eaf6c 100644
--- a/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc
+++ b/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc
@@ -21,7 +21,6 @@
 #include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
 #include "webrtc/modules/audio_coding/acm2/acm_resampler.h"
 #include "webrtc/modules/audio_coding/acm2/call_statistics.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/logging.h"
 #include "webrtc/system_wrappers/include/metrics.h"
 #include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
@@ -103,8 +102,7 @@ void AudioCodingModuleImpl::ChangeLogger::MaybeLog(int value) {
 
 AudioCodingModuleImpl::AudioCodingModuleImpl(
     const AudioCodingModule::Config& config)
-    : acm_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
-      id_(config.id),
+    : id_(config.id),
       expected_codec_ts_(0xD87F3F9F),
       expected_in_ts_(0xD87F3F9F),
       receiver_(config),
@@ -113,7 +111,6 @@ AudioCodingModuleImpl::AudioCodingModuleImpl(
       receiver_initialized_(false),
       first_10ms_data_(false),
       first_frame_(true),
-      callback_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
       packetization_callback_(NULL),
       vad_callback_(NULL) {
   if (InitializeReceiverSafe() < 0) {
@@ -173,7 +170,7 @@ int32_t AudioCodingModuleImpl::Encode(const InputData& input_data) {
   }
 
   {
-    CriticalSectionScoped lock(callback_crit_sect_.get());
+    rtc::CritScope lock(&callback_crit_sect_);
     if (packetization_callback_) {
       packetization_callback_->SendData(
           frame_type, encoded_info.payload_type, encoded_info.encoded_timestamp,
@@ -197,7 +194,7 @@ int32_t AudioCodingModuleImpl::Encode(const InputData& input_data) {
 
 // Can be called multiple times for Codec, CNG, RED.
 int AudioCodingModuleImpl::RegisterSendCodec(const CodecInst& send_codec) {
-  CriticalSectionScoped lock(acm_crit_sect_.get());
+  rtc::CritScope lock(&acm_crit_sect_);
   if (!codec_manager_.RegisterEncoder(send_codec)) {
     return -1;
   }
@@ -217,7 +214,7 @@ int AudioCodingModuleImpl::RegisterSendCodec(const CodecInst& send_codec) {
 
 void AudioCodingModuleImpl::RegisterExternalSendCodec(
     AudioEncoder* external_speech_encoder) {
-  CriticalSectionScoped lock(acm_crit_sect_.get());
+  rtc::CritScope lock(&acm_crit_sect_);
   auto* sp = codec_manager_.GetStackParams();
   sp->speech_encoder = external_speech_encoder;
   rent_a_codec_.RentEncoderStack(sp);
@@ -225,7 +222,7 @@ void AudioCodingModuleImpl::RegisterExternalSendCodec(
 
 // Get current send codec.
 rtc::Optional<CodecInst> AudioCodingModuleImpl::SendCodec() const {
-  CriticalSectionScoped lock(acm_crit_sect_.get());
+  rtc::CritScope lock(&acm_crit_sect_);
   auto* ci = codec_manager_.GetCodecInst();
   if (ci) {
     return rtc::Optional<CodecInst>(*ci);
@@ -241,7 +238,7 @@ rtc::Optional<CodecInst> AudioCodingModuleImpl::SendCodec() const {
 int AudioCodingModuleImpl::SendFrequency() const {
   WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_,
                "SendFrequency()");
-  CriticalSectionScoped lock(acm_crit_sect_.get());
+  rtc::CritScope lock(&acm_crit_sect_);
 
   const auto* enc = rent_a_codec_.GetEncoderStack();
   if (!enc) {
@@ -254,7 +251,7 @@ int AudioCodingModuleImpl::SendFrequency() const {
 }
 
 void AudioCodingModuleImpl::SetBitRate(int bitrate_bps) {
-  CriticalSectionScoped lock(acm_crit_sect_.get());
+  rtc::CritScope lock(&acm_crit_sect_);
   auto* enc = rent_a_codec_.GetEncoderStack();
   if (enc) {
     enc->SetTargetBitrate(bitrate_bps);
@@ -265,7 +262,7 @@ void AudioCodingModuleImpl::SetBitRate(int bitrate_bps) {
 // the encoded buffers.
 int AudioCodingModuleImpl::RegisterTransportCallback(
     AudioPacketizationCallback* transport) {
-  CriticalSectionScoped lock(callback_crit_sect_.get());
+  rtc::CritScope lock(&callback_crit_sect_);
   packetization_callback_ = transport;
   return 0;
 }
@@ -273,7 +270,7 @@ int AudioCodingModuleImpl::RegisterTransportCallback(
 // Add 10MS of raw (PCM) audio data to the encoder.
 int AudioCodingModuleImpl::Add10MsData(const AudioFrame& audio_frame) {
   InputData input_data;
-  CriticalSectionScoped lock(acm_crit_sect_.get());
+  rtc::CritScope lock(&acm_crit_sect_);
   int r = Add10MsDataInternal(audio_frame, &input_data);
   return r < 0 ? r : Encode(input_data);
 }
@@ -445,14 +442,14 @@ int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame,
 //
 
 bool AudioCodingModuleImpl::REDStatus() const {
-  CriticalSectionScoped lock(acm_crit_sect_.get());
+  rtc::CritScope lock(&acm_crit_sect_);
   return codec_manager_.GetStackParams()->use_red;
 }
 
 // Configure RED status i.e on/off.
 int AudioCodingModuleImpl::SetREDStatus(bool enable_red) {
 #ifdef WEBRTC_CODEC_RED
-  CriticalSectionScoped lock(acm_crit_sect_.get());
+  rtc::CritScope lock(&acm_crit_sect_);
   if (!codec_manager_.SetCopyRed(enable_red)) {
     return -1;
   }
@@ -472,12 +469,12 @@ int AudioCodingModuleImpl::SetREDStatus(bool enable_red) {
 //
 
 bool AudioCodingModuleImpl::CodecFEC() const {
-  CriticalSectionScoped lock(acm_crit_sect_.get());
+  rtc::CritScope lock(&acm_crit_sect_);
   return codec_manager_.GetStackParams()->use_codec_fec;
 }
 
 int AudioCodingModuleImpl::SetCodecFEC(bool enable_codec_fec) {
-  CriticalSectionScoped lock(acm_crit_sect_.get());
+  rtc::CritScope lock(&acm_crit_sect_);
   if (!codec_manager_.SetCodecFEC(enable_codec_fec)) {
     return -1;
   }
@@ -493,7 +490,7 @@ int AudioCodingModuleImpl::SetCodecFEC(bool enable_codec_fec) {
 }
 
 int AudioCodingModuleImpl::SetPacketLossRate(int loss_rate) {
-  CriticalSectionScoped lock(acm_crit_sect_.get());
+  rtc::CritScope lock(&acm_crit_sect_);
   if (HaveValidEncoder("SetPacketLossRate")) {
     rent_a_codec_.GetEncoderStack()->SetProjectedPacketLossRate(loss_rate /
                                                                 100.0);
@@ -509,7 +506,7 @@ int AudioCodingModuleImpl::SetVAD(bool enable_dtx,
                                   ACMVADMode mode) {
   // Note: |enable_vad| is not used; VAD is enabled based on the DTX setting.
   RTC_DCHECK_EQ(enable_dtx, enable_vad);
-  CriticalSectionScoped lock(acm_crit_sect_.get());
+  rtc::CritScope lock(&acm_crit_sect_);
   if (!codec_manager_.SetVAD(enable_dtx, mode)) {
     return -1;
   }
@@ -522,7 +519,7 @@ int AudioCodingModuleImpl::SetVAD(bool enable_dtx,
 // Get VAD/DTX settings.
 int AudioCodingModuleImpl::VAD(bool* dtx_enabled, bool* vad_enabled,
                                ACMVADMode* mode) const {
-  CriticalSectionScoped lock(acm_crit_sect_.get());
+  rtc::CritScope lock(&acm_crit_sect_);
   const auto* sp = codec_manager_.GetStackParams();
   *dtx_enabled = *vad_enabled = sp->use_cng;
   *mode = sp->vad_mode;
@@ -534,7 +531,7 @@ int AudioCodingModuleImpl::VAD(bool* dtx_enabled, bool* vad_enabled,
 //
 
 int AudioCodingModuleImpl::InitializeReceiver() {
-  CriticalSectionScoped lock(acm_crit_sect_.get());
+  rtc::CritScope lock(&acm_crit_sect_);
   return InitializeReceiverSafe();
 }
 
@@ -587,7 +584,7 @@ int AudioCodingModuleImpl::PlayoutFrequency() const {
 // Register possible receive codecs, can be called multiple times,
 // for codecs, CNG (NB, WB and SWB), DTMF, RED.
 int AudioCodingModuleImpl::RegisterReceiveCodec(const CodecInst& codec) {
-  CriticalSectionScoped lock(acm_crit_sect_.get());
+  rtc::CritScope lock(&acm_crit_sect_);
   RTC_DCHECK(receiver_initialized_);
   if (codec.channels > 2) {
     LOG_F(LS_ERROR) << "Unsupported number of channels: " << codec.channels;
@@ -625,7 +622,7 @@ int AudioCodingModuleImpl::RegisterExternalReceiveCodec(
     int sample_rate_hz,
     int num_channels,
     const std::string& name) {
-  CriticalSectionScoped lock(acm_crit_sect_.get());
+  rtc::CritScope lock(&acm_crit_sect_);
   RTC_DCHECK(receiver_initialized_);
   if (num_channels > 2 || num_channels < 0) {
     LOG_F(LS_ERROR) << "Unsupported number of channels: " << num_channels;
@@ -645,7 +642,7 @@ int AudioCodingModuleImpl::RegisterExternalReceiveCodec(
 
 // Get current received codec.
 int AudioCodingModuleImpl::ReceiveCodec(CodecInst* current_codec) const {
-  CriticalSectionScoped lock(acm_crit_sect_.get());
+  rtc::CritScope lock(&acm_crit_sect_);
   return receiver_.LastAudioCodec(current_codec);
 }
 
@@ -705,7 +702,7 @@ int AudioCodingModuleImpl::GetNetworkStatistics(NetworkStatistics* statistics) {
 int AudioCodingModuleImpl::RegisterVADCallback(ACMVADCallback* vad_callback) {
   WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceAudioCoding, id_,
                "RegisterVADCallback()");
-  CriticalSectionScoped lock(callback_crit_sect_.get());
+  rtc::CritScope lock(&callback_crit_sect_);
   vad_callback_ = vad_callback;
   return 0;
 }
@@ -740,7 +737,7 @@ int AudioCodingModuleImpl::IncomingPayload(const uint8_t* incoming_payload,
 }
 
 int AudioCodingModuleImpl::SetOpusApplication(OpusApplicationMode application) {
-  CriticalSectionScoped lock(acm_crit_sect_.get());
+  rtc::CritScope lock(&acm_crit_sect_);
   if (!HaveValidEncoder("SetOpusApplication")) {
     return -1;
   }
@@ -761,7 +758,7 @@ int AudioCodingModuleImpl::SetOpusApplication(OpusApplicationMode application) {
 
 // Informs Opus encoder of the maximum playback rate the receiver will render.
 int AudioCodingModuleImpl::SetOpusMaxPlaybackRate(int frequency_hz) {
-  CriticalSectionScoped lock(acm_crit_sect_.get());
+  rtc::CritScope lock(&acm_crit_sect_);
   if (!HaveValidEncoder("SetOpusMaxPlaybackRate")) {
     return -1;
   }
@@ -770,7 +767,7 @@ int AudioCodingModuleImpl::SetOpusMaxPlaybackRate(int frequency_hz) {
 }
 
 int AudioCodingModuleImpl::EnableOpusDtx() {
-  CriticalSectionScoped lock(acm_crit_sect_.get());
+  rtc::CritScope lock(&acm_crit_sect_);
   if (!HaveValidEncoder("EnableOpusDtx")) {
     return -1;
   }
@@ -778,7 +775,7 @@ int AudioCodingModuleImpl::EnableOpusDtx() {
 }
 
 int AudioCodingModuleImpl::DisableOpusDtx() {
-  CriticalSectionScoped lock(acm_crit_sect_.get());
+  rtc::CritScope lock(&acm_crit_sect_);
   if (!HaveValidEncoder("DisableOpusDtx")) {
     return -1;
   }
diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h b/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h
index 926671f199..19776d1013 100644
--- a/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h
+++ b/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h
@@ -15,6 +15,7 @@
 #include <vector>
 
 #include "webrtc/base/buffer.h"
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/common_types.h"
@@ -25,7 +26,6 @@
 
 namespace webrtc {
 
-class CriticalSectionWrapper;
 class AudioCodingImpl;
 
 namespace acm2 {
@@ -240,7 +240,7 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
   // to |index|.
   int UpdateUponReceivingCodec(int index);
 
-  const rtc::scoped_ptr<CriticalSectionWrapper> acm_crit_sect_;
+  mutable rtc::CriticalSection acm_crit_sect_;
   rtc::Buffer encode_buffer_ GUARDED_BY(acm_crit_sect_);
   int id_;  // TODO(henrik.lundin) Make const.
   uint32_t expected_codec_ts_ GUARDED_BY(acm_crit_sect_);
@@ -271,7 +271,7 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
   uint32_t last_timestamp_ GUARDED_BY(acm_crit_sect_);
   uint32_t last_rtp_timestamp_ GUARDED_BY(acm_crit_sect_);
 
-  const rtc::scoped_ptr<CriticalSectionWrapper> callback_crit_sect_;
+  mutable rtc::CriticalSection callback_crit_sect_;
   AudioPacketizationCallback* packetization_callback_
       GUARDED_BY(callback_crit_sect_);
   ACMVADCallback* vad_callback_ GUARDED_BY(callback_crit_sect_);
diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc b/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
index 6f82a96ee5..c738d0f095 100644
--- a/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
+++ b/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
@@ -13,17 +13,18 @@
 #include <vector>
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/base/md5digest.h"
 #include "webrtc/base/platform_thread.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
+#include "webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.h"
+#include "webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
 #include "webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
 #include "webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
 #include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
 #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h"
-#include "webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.h"
-#include "webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
 #include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h"
@@ -37,7 +38,6 @@
 #include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/clock.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/test/testsupport/fileutils.h"
@@ -94,8 +94,7 @@ class PacketizationCallbackStubOldApi : public AudioPacketizationCallback {
       : num_calls_(0),
         last_frame_type_(kEmptyFrame),
         last_payload_type_(-1),
-        last_timestamp_(0),
-        crit_sect_(CriticalSectionWrapper::CreateCriticalSection()) {}
+        last_timestamp_(0) {}
 
   int32_t SendData(FrameType frame_type,
                    uint8_t payload_type,
@@ -103,7 +102,7 @@ class PacketizationCallbackStubOldApi : public AudioPacketizationCallback {
                    const uint8_t* payload_data,
                    size_t payload_len_bytes,
                    const RTPFragmentationHeader* fragmentation) override {
-    CriticalSectionScoped lock(crit_sect_.get());
+    rtc::CritScope lock(&crit_sect_);
     ++num_calls_;
     last_frame_type_ = frame_type;
     last_payload_type_ = payload_type;
@@ -113,32 +112,32 @@ class PacketizationCallbackStubOldApi : public AudioPacketizationCallback {
   }
 
   int num_calls() const {
-    CriticalSectionScoped lock(crit_sect_.get());
+    rtc::CritScope lock(&crit_sect_);
     return num_calls_;
   }
 
   int last_payload_len_bytes() const {
-    CriticalSectionScoped lock(crit_sect_.get());
+    rtc::CritScope lock(&crit_sect_);
     return last_payload_vec_.size();
   }
 
   FrameType last_frame_type() const {
-    CriticalSectionScoped lock(crit_sect_.get());
+    rtc::CritScope lock(&crit_sect_);
     return last_frame_type_;
   }
 
   int last_payload_type() const {
-    CriticalSectionScoped lock(crit_sect_.get());
+    rtc::CritScope lock(&crit_sect_);
     return last_payload_type_;
   }
 
   uint32_t last_timestamp() const {
-    CriticalSectionScoped lock(crit_sect_.get());
+    rtc::CritScope lock(&crit_sect_);
     return last_timestamp_;
   }
 
   void SwapBuffers(std::vector<uint8_t>* payload) {
-    CriticalSectionScoped lock(crit_sect_.get());
+    rtc::CritScope lock(&crit_sect_);
     last_payload_vec_.swap(*payload);
   }
 
@@ -148,7 +147,7 @@ class PacketizationCallbackStubOldApi : public AudioPacketizationCallback {
   int last_payload_type_ GUARDED_BY(crit_sect_);
   uint32_t last_timestamp_ GUARDED_BY(crit_sect_);
   std::vector<uint8_t> last_payload_vec_ GUARDED_BY(crit_sect_);
-  const rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
+  mutable rtc::CriticalSection crit_sect_;
 };
 
 class AudioCodingModuleTestOldApi : public ::testing::Test {
@@ -469,7 +468,6 @@ class AudioCodingModuleMtTestOldApi : public AudioCodingModuleTestOldApi {
         send_count_(0),
         insert_packet_count_(0),
         pull_audio_count_(0),
-        crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
         next_insert_packet_time_ms_(0),
         fake_clock_(new SimulatedClock(0)) {
     clock_ = fake_clock_.get();
@@ -503,7 +501,7 @@ class AudioCodingModuleMtTestOldApi : public AudioCodingModuleTestOldApi {
 
   virtual bool TestDone() {
     if (packet_cb_.num_calls() > kNumPackets) {
-      CriticalSectionScoped lock(crit_sect_.get());
+      rtc::CritScope lock(&crit_sect_);
       if (pull_audio_count_ > kNumPullCalls) {
         // Both conditions for completion are met. End the test.
         return true;
@@ -541,7 +539,7 @@ class AudioCodingModuleMtTestOldApi : public AudioCodingModuleTestOldApi {
   bool CbInsertPacketImpl() {
     SleepMs(1);
     {
-      CriticalSectionScoped lock(crit_sect_.get());
+      rtc::CritScope lock(&crit_sect_);
       if (clock_->TimeInMilliseconds() < next_insert_packet_time_ms_) {
         return true;
       }
@@ -561,7 +559,7 @@ class AudioCodingModuleMtTestOldApi : public AudioCodingModuleTestOldApi {
   bool CbPullAudioImpl() {
     SleepMs(1);
     {
-      CriticalSectionScoped lock(crit_sect_.get());
+      rtc::CritScope lock(&crit_sect_);
       // Don't let the insert thread fall behind.
       if (next_insert_packet_time_ms_ < clock_->TimeInMilliseconds()) {
         return true;
@@ -581,7 +579,7 @@ class AudioCodingModuleMtTestOldApi : public AudioCodingModuleTestOldApi {
   int send_count_;
   int insert_packet_count_;
   int pull_audio_count_ GUARDED_BY(crit_sect_);
-  const rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
+  mutable rtc::CriticalSection crit_sect_;
   int64_t next_insert_packet_time_ms_ GUARDED_BY(crit_sect_);
   rtc::scoped_ptr<SimulatedClock> fake_clock_;
 };
@@ -681,7 +679,7 @@ class AcmIsacMtTestOldApi : public AudioCodingModuleMtTestOldApi {
   // run).
   virtual bool TestDone() {
     if (packet_cb_.num_calls() > kNumPackets) {
-      CriticalSectionScoped lock(crit_sect_.get());
+      rtc::CritScope lock(&crit_sect_);
       if (pull_audio_count_ > kNumPullCalls) {
         // Both conditions for completion are met. End the test.
         return true;
@@ -720,7 +718,6 @@ class AcmReRegisterIsacMtTestOldApi : public AudioCodingModuleTestOldApi {
                                    this,
                                    "codec_registration"),
         test_complete_(EventWrapper::Create()),
-        crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
         codec_registered_(false),
         receive_packet_count_(0),
         next_insert_packet_time_ms_(0),
@@ -781,7 +778,7 @@ class AcmReRegisterIsacMtTestOldApi : public AudioCodingModuleTestOldApi {
     rtc::scoped_ptr<uint8_t[]> encoded(new uint8_t[max_encoded_bytes]);
     AudioEncoder::EncodedInfo info;
     {
-      CriticalSectionScoped lock(crit_sect_.get());
+      rtc::CritScope lock(&crit_sect_);
       if (clock_->TimeInMilliseconds() < next_insert_packet_time_ms_) {
         return true;
       }
@@ -829,7 +826,7 @@ class AcmReRegisterIsacMtTestOldApi : public AudioCodingModuleTestOldApi {
       // End the test early if a fatal failure (ASSERT_*) has occurred.
       test_complete_->Set();
     }
-    CriticalSectionScoped lock(crit_sect_.get());
+    rtc::CritScope lock(&crit_sect_);
     if (!codec_registered_ &&
         receive_packet_count_ > kRegisterAfterNumPackets) {
       // Register the iSAC encoder.
@@ -845,7 +842,7 @@ class AcmReRegisterIsacMtTestOldApi : public AudioCodingModuleTestOldApi {
   rtc::PlatformThread receive_thread_;
   rtc::PlatformThread codec_registration_thread_;
   const rtc::scoped_ptr<EventWrapper> test_complete_;
-  const rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
+  mutable rtc::CriticalSection crit_sect_;
   bool codec_registered_ GUARDED_BY(crit_sect_);
   int receive_packet_count_ GUARDED_BY(crit_sect_);
   int64_t next_insert_packet_time_ms_ GUARDED_BY(crit_sect_);
diff --git a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
index d4438cc775..b9da7f49e9 100644
--- a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
+++ b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
@@ -14,6 +14,7 @@
 #include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
 
 #include "webrtc/base/checks.h"
+#include "webrtc/common_types.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.cc b/webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.cc
index 78b415c4c9..86f60555f7 100644
--- a/webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.cc
+++ b/webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.cc
@@ -12,8 +12,7 @@
 
 namespace webrtc {
 
-LockedIsacBandwidthInfo::LockedIsacBandwidthInfo()
-      : lock_(CriticalSectionWrapper::CreateCriticalSection()) {
+LockedIsacBandwidthInfo::LockedIsacBandwidthInfo() {
   bwinfo_.in_use = 0;
 }
 
diff --git a/webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h b/webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h
index bbb040de1d..9996cbdbef 100644
--- a/webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h
+++ b/webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h
@@ -11,10 +11,10 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_LOCKED_BANDWIDTH_INFO_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_LOCKED_BANDWIDTH_INFO_H_
 
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/modules/audio_coding/codecs/isac/bandwidth_info.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
 
@@ -26,17 +26,17 @@ class LockedIsacBandwidthInfo final {
   ~LockedIsacBandwidthInfo();
 
   IsacBandwidthInfo Get() const {
-    CriticalSectionScoped cs(lock_.get());
+    rtc::CritScope lock(&lock_);
     return bwinfo_;
   }
 
   void Set(const IsacBandwidthInfo& bwinfo) {
-    CriticalSectionScoped cs(lock_.get());
+    rtc::CritScope lock(&lock_);
     bwinfo_ = bwinfo;
   }
 
  private:
-  const rtc::scoped_ptr<CriticalSectionWrapper> lock_;
+  mutable rtc::CriticalSection lock_;
   IsacBandwidthInfo bwinfo_ GUARDED_BY(lock_);
 };
 
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.cc b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
index 6c07da46f0..93699821cf 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
@@ -44,7 +44,6 @@
 #include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
 #include "webrtc/modules/audio_coding/neteq/timestamp_scaler.h"
 #include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 // Modify the code to obtain backwards bit-exactness. Once bit-exactness is no
 // longer required, this #define should be removed (and the code that it
@@ -67,8 +66,7 @@ NetEqImpl::NetEqImpl(const NetEq::Config& config,
                      ExpandFactory* expand_factory,
                      PreemptiveExpandFactory* preemptive_expand_factory,
                      bool create_components)
-    : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
-      buffer_level_filter_(buffer_level_filter),
+    : buffer_level_filter_(buffer_level_filter),
       decoder_database_(decoder_database),
       delay_manager_(delay_manager),
       delay_peak_detector_(delay_peak_detector),
@@ -126,7 +124,7 @@ int NetEqImpl::InsertPacket(const WebRtcRTPHeader& rtp_header,
                             rtc::ArrayView<const uint8_t> payload,
                             uint32_t receive_timestamp) {
   TRACE_EVENT0("webrtc", "NetEqImpl::InsertPacket");
-  CriticalSectionScoped lock(crit_sect_.get());
+  rtc::CritScope lock(&crit_sect_);
   int error =
       InsertPacketInternal(rtp_header, payload, receive_timestamp, false);
   if (error != 0) {
@@ -138,7 +136,7 @@ int NetEqImpl::InsertPacket(const WebRtcRTPHeader& rtp_header,
 
 int NetEqImpl::InsertSyncPacket(const WebRtcRTPHeader& rtp_header,
                                 uint32_t receive_timestamp) {
-  CriticalSectionScoped lock(crit_sect_.get());
+  rtc::CritScope lock(&crit_sect_);
   const uint8_t kSyncPayload[] = { 's', 'y', 'n', 'c' };
   int error =
       InsertPacketInternal(rtp_header, kSyncPayload, receive_timestamp, true);
@@ -154,7 +152,7 @@ int NetEqImpl::GetAudio(size_t max_length, int16_t* output_audio,
                         size_t* samples_per_channel, size_t* num_channels,
                         NetEqOutputType* type) {
   TRACE_EVENT0("webrtc", "NetEqImpl::GetAudio");
-  CriticalSectionScoped lock(crit_sect_.get());
+  rtc::CritScope lock(&crit_sect_);
   int error = GetAudioInternal(max_length, output_audio, samples_per_channel,
                                num_channels);
   if (error != 0) {
@@ -177,7 +175,7 @@ int NetEqImpl::GetAudio(size_t max_length, int16_t* output_audio,
 int NetEqImpl::RegisterPayloadType(NetEqDecoder codec,
                                    const std::string& name,
                                    uint8_t rtp_payload_type) {
-  CriticalSectionScoped lock(crit_sect_.get());
+  rtc::CritScope lock(&crit_sect_);
   LOG(LS_VERBOSE) << "RegisterPayloadType "
                   << static_cast<int>(rtp_payload_type) << " "
                   << static_cast<int>(codec);
@@ -206,7 +204,7 @@ int NetEqImpl::RegisterExternalDecoder(AudioDecoder* decoder,
                                        const std::string& codec_name,
                                        uint8_t rtp_payload_type,
                                        int sample_rate_hz) {
-  CriticalSectionScoped lock(crit_sect_.get());
+  rtc::CritScope lock(&crit_sect_);
   LOG(LS_VERBOSE) << "RegisterExternalDecoder "
                   << static_cast<int>(rtp_payload_type) << " "
                   << static_cast<int>(codec);
@@ -243,7 +241,7 @@ int NetEqImpl::RegisterExternalDecoder(AudioDecoder* decoder,
 }
 
 int NetEqImpl::RemovePayloadType(uint8_t rtp_payload_type) {
-  CriticalSectionScoped lock(crit_sect_.get());
+  rtc::CritScope lock(&crit_sect_);
   int ret = decoder_database_->Remove(rtp_payload_type);
   if (ret == DecoderDatabase::kOK) {
     return kOK;
@@ -256,7 +254,7 @@ int NetEqImpl::RemovePayloadType(uint8_t rtp_payload_type) {
 }
 
 bool NetEqImpl::SetMinimumDelay(int delay_ms) {
-  CriticalSectionScoped lock(crit_sect_.get());
+  rtc::CritScope lock(&crit_sect_);
   if (delay_ms >= 0 && delay_ms < 10000) {
     assert(delay_manager_.get());
     return delay_manager_->SetMinimumDelay(delay_ms);
@@ -265,7 +263,7 @@ bool NetEqImpl::SetMinimumDelay(int delay_ms) {
 }
 
 bool NetEqImpl::SetMaximumDelay(int delay_ms) {
-  CriticalSectionScoped lock(crit_sect_.get());
+  rtc::CritScope lock(&crit_sect_);
   if (delay_ms >= 0 && delay_ms < 10000) {
     assert(delay_manager_.get());
     return delay_manager_->SetMaximumDelay(delay_ms);
@@ -274,7 +272,7 @@ bool NetEqImpl::SetMaximumDelay(int delay_ms) {
 }
 
 int NetEqImpl::LeastRequiredDelayMs() const {
-  CriticalSectionScoped lock(crit_sect_.get());
+  rtc::CritScope lock(&crit_sect_);
   assert(delay_manager_.get());
   return delay_manager_->least_required_delay_ms();
 }
@@ -288,7 +286,7 @@ int NetEqImpl::TargetDelay() {
 }
 
 int NetEqImpl::CurrentDelayMs() const {
-  CriticalSectionScoped lock(crit_sect_.get());
+  rtc::CritScope lock(&crit_sect_);
   if (fs_hz_ == 0)
     return 0;
   // Sum up the samples in the packet buffer with the future length of the sync
@@ -306,7 +304,7 @@ int NetEqImpl::CurrentDelayMs() const {
 // Deprecated.
 // TODO(henrik.lundin) Delete.
 void NetEqImpl::SetPlayoutMode(NetEqPlayoutMode mode) {
-  CriticalSectionScoped lock(crit_sect_.get());
+  rtc::CritScope lock(&crit_sect_);
   if (mode != playout_mode_) {
     playout_mode_ = mode;
     CreateDecisionLogic();
@@ -316,12 +314,12 @@ void NetEqImpl::SetPlayoutMode(NetEqPlayoutMode mode) {
 // Deprecated.
 // TODO(henrik.lundin) Delete.
 NetEqPlayoutMode NetEqImpl::PlayoutMode() const {
-  CriticalSectionScoped lock(crit_sect_.get());
+  rtc::CritScope lock(&crit_sect_);
   return playout_mode_;
 }
 
 int NetEqImpl::NetworkStatistics(NetEqNetworkStatistics* stats) {
-  CriticalSectionScoped lock(crit_sect_.get());
+  rtc::CritScope lock(&crit_sect_);
   assert(decoder_database_.get());
   const size_t total_samples_in_buffers =
       packet_buffer_->NumSamplesInBuffer(decoder_database_.get(),
@@ -336,33 +334,33 @@ int NetEqImpl::NetworkStatistics(NetEqNetworkStatistics* stats) {
 }
 
 void NetEqImpl::GetRtcpStatistics(RtcpStatistics* stats) {
-  CriticalSectionScoped lock(crit_sect_.get());
+  rtc::CritScope lock(&crit_sect_);
   if (stats) {
     rtcp_.GetStatistics(false, stats);
   }
 }
 
 void NetEqImpl::GetRtcpStatisticsNoReset(RtcpStatistics* stats) {
-  CriticalSectionScoped lock(crit_sect_.get());
+  rtc::CritScope lock(&crit_sect_);
   if (stats) {
     rtcp_.GetStatistics(true, stats);
   }
 }
 
 void NetEqImpl::EnableVad() {
-  CriticalSectionScoped lock(crit_sect_.get());
+  rtc::CritScope lock(&crit_sect_);
   assert(vad_.get());
   vad_->Enable();
 }
 
 void NetEqImpl::DisableVad() {
-  CriticalSectionScoped lock(crit_sect_.get());
+  rtc::CritScope lock(&crit_sect_);
   assert(vad_.get());
   vad_->Disable();
 }
 
 bool NetEqImpl::GetPlayoutTimestamp(uint32_t* timestamp) {
-  CriticalSectionScoped lock(crit_sect_.get());
+  rtc::CritScope lock(&crit_sect_);
   if (first_packet_) {
     // We don't have a valid RTP timestamp until we have decoded our first
     // RTP packet.
@@ -373,7 +371,7 @@ bool NetEqImpl::GetPlayoutTimestamp(uint32_t* timestamp) {
 }
 
 int NetEqImpl::last_output_sample_rate_hz() const {
-  CriticalSectionScoped lock(crit_sect_.get());
+  rtc::CritScope lock(&crit_sect_);
   return last_output_sample_rate_hz_;
 }
 
@@ -386,17 +384,17 @@ int NetEqImpl::SetTargetSampleRate() {
 }
 
 int NetEqImpl::LastError() const {
-  CriticalSectionScoped lock(crit_sect_.get());
+  rtc::CritScope lock(&crit_sect_);
   return error_code_;
 }
 
 int NetEqImpl::LastDecoderError() {
-  CriticalSectionScoped lock(crit_sect_.get());
+  rtc::CritScope lock(&crit_sect_);
   return decoder_error_code_;
 }
 
 void NetEqImpl::FlushBuffers() {
-  CriticalSectionScoped lock(crit_sect_.get());
+  rtc::CritScope lock(&crit_sect_);
   LOG(LS_VERBOSE) << "FlushBuffers";
   packet_buffer_->Flush();
   assert(sync_buffer_.get());
@@ -410,12 +408,12 @@ void NetEqImpl::FlushBuffers() {
 
 void NetEqImpl::PacketBufferStatistics(int* current_num_packets,
                                        int* max_num_packets) const {
-  CriticalSectionScoped lock(crit_sect_.get());
+  rtc::CritScope lock(&crit_sect_);
   packet_buffer_->BufferStat(current_num_packets, max_num_packets);
 }
 
 void NetEqImpl::EnableNack(size_t max_nack_list_size) {
-  CriticalSectionScoped lock(crit_sect_.get());
+  rtc::CritScope lock(&crit_sect_);
   if (!nack_enabled_) {
     const int kNackThresholdPackets = 2;
     nack_.reset(Nack::Create(kNackThresholdPackets));
@@ -426,13 +424,13 @@ void NetEqImpl::EnableNack(size_t max_nack_list_size) {
 }
 
 void NetEqImpl::DisableNack() {
-  CriticalSectionScoped lock(crit_sect_.get());
+  rtc::CritScope lock(&crit_sect_);
   nack_.reset();
   nack_enabled_ = false;
 }
 
 std::vector<uint16_t> NetEqImpl::GetNackList(int64_t round_trip_time_ms) const {
-  CriticalSectionScoped lock(crit_sect_.get());
+  rtc::CritScope lock(&crit_sect_);
   if (!nack_enabled_) {
     return std::vector<uint16_t>();
   }
@@ -441,7 +439,7 @@ std::vector<uint16_t> NetEqImpl::GetNackList(int64_t round_trip_time_ms) const {
 }
 
 const SyncBuffer* NetEqImpl::sync_buffer_for_test() const {
-  CriticalSectionScoped lock(crit_sect_.get());
+  rtc::CritScope lock(&crit_sect_);
   return sync_buffer_.get();
 }
 
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.h b/webrtc/modules/audio_coding/neteq/neteq_impl.h
index 940deadd2f..817b697bcf 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.h
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.h
@@ -14,6 +14,7 @@
 #include <string>
 
 #include "webrtc/base/constructormagic.h"
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
@@ -32,7 +33,6 @@ class Accelerate;
 class BackgroundNoise;
 class BufferLevelFilter;
 class ComfortNoise;
-class CriticalSectionWrapper;
 class DecisionLogic;
 class DecoderDatabase;
 class DelayManager;
@@ -338,7 +338,7 @@ class NetEqImpl : public webrtc::NetEq {
   // Creates DecisionLogic object with the mode given by |playout_mode_|.
   virtual void CreateDecisionLogic() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
 
-  const rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
+  mutable rtc::CriticalSection crit_sect_;
   const rtc::scoped_ptr<BufferLevelFilter> buffer_level_filter_
       GUARDED_BY(crit_sect_);
   const rtc::scoped_ptr<DecoderDatabase> decoder_database_
diff --git a/webrtc/modules/audio_coding/test/Channel.cc b/webrtc/modules/audio_coding/test/Channel.cc
index 31521fe1e3..0507691fb4 100644
--- a/webrtc/modules/audio_coding/test/Channel.cc
+++ b/webrtc/modules/audio_coding/test/Channel.cc
@@ -15,7 +15,6 @@
 
 #include "webrtc/base/format_macros.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
 
@@ -95,7 +94,7 @@ int32_t Channel::SendData(FrameType frameType,
     }
   }
 
-  _channelCritSect->Enter();
+  _channelCritSect.Enter();
   if (_saveBitStream) {
     //fwrite(payloadData, sizeof(uint8_t), payloadSize, _bitStreamFile);
   }
@@ -106,7 +105,7 @@ int32_t Channel::SendData(FrameType frameType,
   _useLastFrameSize = false;
   _lastInTimestamp = timeStamp;
   _totalBytes += payloadDataSize;
-  _channelCritSect->Leave();
+  _channelCritSect.Leave();
 
   if (_useFECTestWithPacketLoss) {
     _packetLoss += 1;
@@ -225,7 +224,6 @@ void Channel::CalcStatistics(WebRtcRTPHeader& rtpInfo, size_t payloadSize) {
 Channel::Channel(int16_t chID)
     : _receiverACM(NULL),
       _seqNo(0),
-      _channelCritSect(CriticalSectionWrapper::CreateCriticalSection()),
       _bitStreamFile(NULL),
       _saveBitStream(false),
       _lastPayloadType(-1),
@@ -265,7 +263,6 @@ Channel::Channel(int16_t chID)
 }
 
 Channel::~Channel() {
-  delete _channelCritSect;
 }
 
 void Channel::RegisterReceiverACM(AudioCodingModule* acm) {
@@ -276,7 +273,7 @@ void Channel::RegisterReceiverACM(AudioCodingModule* acm) {
 void Channel::ResetStats() {
   int n;
   int k;
-  _channelCritSect->Enter();
+  _channelCritSect.Enter();
   _lastPayloadType = -1;
   for (n = 0; n < MAX_NUM_PAYLOADS; n++) {
     _payloadStats[n].payloadType = -1;
@@ -291,12 +288,12 @@ void Channel::ResetStats() {
   }
   _beginTime = TickTime::MillisecondTimestamp();
   _totalBytes = 0;
-  _channelCritSect->Leave();
+  _channelCritSect.Leave();
 }
 
 int16_t Channel::Stats(CodecInst& codecInst,
                        ACMTestPayloadStats& payloadStats) {
-  _channelCritSect->Enter();
+  _channelCritSect.Enter();
   int n;
   payloadStats.payloadType = -1;
   for (n = 0; n < MAX_NUM_PAYLOADS; n++) {
@@ -306,12 +303,12 @@ int16_t Channel::Stats(CodecInst& codecInst,
     }
   }
   if (payloadStats.payloadType == -1) {
-    _channelCritSect->Leave();
+    _channelCritSect.Leave();
     return -1;
   }
   for (n = 0; n < MAX_NUM_FRAMESIZES; n++) {
     if (payloadStats.frameSizeStats[n].frameSizeSample == 0) {
-      _channelCritSect->Leave();
+      _channelCritSect.Leave();
       return 0;
     }
     payloadStats.frameSizeStats[n].usageLenSec = (double) payloadStats
@@ -322,12 +319,12 @@ int16_t Channel::Stats(CodecInst& codecInst,
             / payloadStats.frameSizeStats[n].usageLenSec;
 
   }
-  _channelCritSect->Leave();
+  _channelCritSect.Leave();
   return 0;
 }
 
 void Channel::Stats(uint32_t* numPackets) {
-  _channelCritSect->Enter();
+  _channelCritSect.Enter();
   int k;
   int n;
   memset(numPackets, 0, MAX_NUM_PAYLOADS * sizeof(uint32_t));
@@ -343,11 +340,11 @@ void Channel::Stats(uint32_t* numPackets) {
       numPackets[k] += _payloadStats[k].frameSizeStats[n].numPackets;
     }
   }
-  _channelCritSect->Leave();
+  _channelCritSect.Leave();
 }
 
 void Channel::Stats(uint8_t* payloadType, uint32_t* payloadLenByte) {
-  _channelCritSect->Enter();
+  _channelCritSect.Enter();
 
   int k;
   int n;
@@ -367,7 +364,7 @@ void Channel::Stats(uint8_t* payloadType, uint32_t* payloadLenByte) {
     }
   }
 
-  _channelCritSect->Leave();
+  _channelCritSect.Leave();
 }
 
 void Channel::PrintStats(CodecInst& codecInst) {
@@ -406,18 +403,18 @@ void Channel::PrintStats(CodecInst& codecInst) {
 
 uint32_t Channel::LastInTimestamp() {
   uint32_t timestamp;
-  _channelCritSect->Enter();
+  _channelCritSect.Enter();
   timestamp = _lastInTimestamp;
-  _channelCritSect->Leave();
+  _channelCritSect.Leave();
   return timestamp;
 }
 
 double Channel::BitRate() {
   double rate;
   uint64_t currTime = TickTime::MillisecondTimestamp();
-  _channelCritSect->Enter();
+  _channelCritSect.Enter();
   rate = ((double) _totalBytes * 8.0) / (double) (currTime - _beginTime);
-  _channelCritSect->Leave();
+  _channelCritSect.Leave();
   return rate;
 }
 
diff --git a/webrtc/modules/audio_coding/test/Channel.h b/webrtc/modules/audio_coding/test/Channel.h
index b047aa9909..3dcd499c03 100644
--- a/webrtc/modules/audio_coding/test/Channel.h
+++ b/webrtc/modules/audio_coding/test/Channel.h
@@ -13,14 +13,13 @@
 
 #include <stdio.h>
 
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
-class CriticalSectionWrapper;
-
 #define MAX_NUM_PAYLOADS   50
 #define MAX_NUM_FRAMESIZES  6
 
@@ -101,7 +100,7 @@ class Channel : public AudioPacketizationCallback {
   // 60msec * 32 sample(max)/msec * 2 description (maybe) * 2 bytes/sample
   uint8_t _payloadData[60 * 32 * 2 * 2];
 
-  CriticalSectionWrapper* _channelCritSect;
+  mutable rtc::CriticalSection _channelCritSect;
   FILE* _bitStreamFile;
   bool _saveBitStream;
   int16_t _lastPayloadType;