Change CriticalSectionScoped to use pointer constructor

BUG=issue183
TEST=audio_coding_module_test

Review URL: https://webrtc-codereview.appspot.com/638005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2384 4adac7df-926f-26a2-2b94-8c16560cd09d

src/modules/audio_coding/main/source/acm_neteq.cc

@@ -61,7 +61,7 @@ _callbackCritSect(CriticalSectionWrapper::CreateCriticalSection())
 
 ACMNetEQ::~ACMNetEQ() {
   {
-    CriticalSectionScoped lock(*_netEqCritSect);
+    CriticalSectionScoped lock(_netEqCritSect);
     RemoveNetEQSafe(0);  // Master.
     RemoveSlavesSafe();
   }
@@ -81,7 +81,7 @@ ACMNetEQ::~ACMNetEQ() {
 WebRtc_Word32
 ACMNetEQ::Init()
 {
-    CriticalSectionScoped lock(*_netEqCritSect);
+    CriticalSectionScoped lock(_netEqCritSect);
 
     for(WebRtc_Word16 idx = 0; idx < _numSlaves + 1; idx++)
     {
@@ -227,7 +227,7 @@ ACMNetEQ::AllocatePacketBuffer(
     // the following has to be int otherwise we will have compiler error
     // if not casted
 
-    CriticalSectionScoped lock(*_netEqCritSect);
+    CriticalSectionScoped lock(_netEqCritSect);
     for(WebRtc_Word16 idx = 0; idx < _numSlaves + 1; idx++)
     {
         if(AllocatePacketBufferByIdxSafe(usedCodecs, noOfCodecs, idx) < 0)
@@ -295,7 +295,7 @@ WebRtc_Word32
 ACMNetEQ::SetExtraDelay(
     const WebRtc_Word32 delayInMS)
 {
-    CriticalSectionScoped lock(*_netEqCritSect);
+    CriticalSectionScoped lock(_netEqCritSect);
 
     for(WebRtc_Word16 idx = 0; idx < _numSlaves + 1; idx++)
     {
@@ -320,7 +320,7 @@ WebRtc_Word32
 ACMNetEQ::SetAVTPlayout(
     const bool enable)
 {
-    CriticalSectionScoped lock(*_netEqCritSect);
+    CriticalSectionScoped lock(_netEqCritSect);
     if (_avtPlayout != enable)
     {
         for(WebRtc_Word16 idx = 0; idx < _numSlaves + 1; idx++)
@@ -346,14 +346,14 @@ ACMNetEQ::SetAVTPlayout(
 bool
 ACMNetEQ::AVTPlayout() const
 {
-    CriticalSectionScoped lock(*_netEqCritSect);
+    CriticalSectionScoped lock(_netEqCritSect);
     return _avtPlayout;
 }
 
 WebRtc_Word32
 ACMNetEQ::CurrentSampFreqHz() const
 {
-    CriticalSectionScoped lock(*_netEqCritSect);
+    CriticalSectionScoped lock(_netEqCritSect);
     if(!_isInitialized[0])
     {
         WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
@@ -368,7 +368,7 @@ WebRtc_Word32
 ACMNetEQ::SetPlayoutMode(
     const AudioPlayoutMode mode)
 {
-    CriticalSectionScoped lock(*_netEqCritSect);
+    CriticalSectionScoped lock(_netEqCritSect);
     if(_playoutMode != mode)
     {
         for(WebRtc_Word16 idx = 0; idx < _numSlaves + 1; idx++)
@@ -408,7 +408,7 @@ ACMNetEQ::SetPlayoutMode(
 AudioPlayoutMode
 ACMNetEQ::PlayoutMode() const
 {
-    CriticalSectionScoped lock(*_netEqCritSect);
+    CriticalSectionScoped lock(_netEqCritSect);
     return _playoutMode;
 }
 
@@ -418,7 +418,7 @@ ACMNetEQ::NetworkStatistics(
     ACMNetworkStatistics* statistics) const
 {
     WebRtcNetEQ_NetworkStatistics stats;
-    CriticalSectionScoped lock(*_netEqCritSect);
+    CriticalSectionScoped lock(_netEqCritSect);
     if(!_isInitialized[0])
     {
         WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
@@ -502,7 +502,7 @@ ACMNetEQ::RecIn(
     netEqRTPInfo.SSRC = rtpInfo.header.ssrc;
     netEqRTPInfo.markerBit = rtpInfo.header.markerBit;
 
-    CriticalSectionScoped lock(*_netEqCritSect);
+    CriticalSectionScoped lock(_netEqCritSect);
     // Down-cast the time to (32-6)-bit since we only care about
     // the least significant bits. (32-6) bits cover 2^(32-6) = 67108864 ms.
     // we masked 6 most significant bits of 32-bit so we don't loose resolution
@@ -574,7 +574,7 @@ ACMNetEQ::RecOut(
 
     WebRtc_Word16 payloadLenSampleSlave;
 
-    CriticalSectionScoped lockNetEq(*_netEqCritSect);
+    CriticalSectionScoped lockNetEq(_netEqCritSect);
 
     if(!_receivedStereo)
     {
@@ -766,7 +766,7 @@ ACMNetEQ::AddCodec(
             "ACMNetEQ::AddCodec: error, codecDef is NULL");
         return -1;
     }
-    CriticalSectionScoped lock(*_netEqCritSect);
+    CriticalSectionScoped lock(_netEqCritSect);
 
     WebRtc_Word16 idx;
     if(toMaster)
@@ -869,7 +869,7 @@ ACMNetEQ::RTPPack(
 WebRtc_Word16
 ACMNetEQ::EnableVAD()
 {
-    CriticalSectionScoped lock(*_netEqCritSect);
+    CriticalSectionScoped lock(_netEqCritSect);
     if (_vadStatus)
     {
         return 0;
@@ -899,7 +899,7 @@ ACMNetEQ::EnableVAD()
 ACMVADMode
 ACMNetEQ::VADMode() const
 {
-    CriticalSectionScoped lock(*_netEqCritSect);
+    CriticalSectionScoped lock(_netEqCritSect);
     return _vadMode;
 }
 
@@ -908,7 +908,7 @@ WebRtc_Word16
 ACMNetEQ::SetVADMode(
     const ACMVADMode mode)
 {
-    CriticalSectionScoped lock(*_netEqCritSect);
+    CriticalSectionScoped lock(_netEqCritSect);
     if((mode < VADNormal) || (mode > VADVeryAggr))
     {
         WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
@@ -940,7 +940,7 @@ ACMNetEQ::SetVADMode(
 WebRtc_Word32
 ACMNetEQ::FlushBuffers()
 {
-    CriticalSectionScoped lock(*_netEqCritSect);
+    CriticalSectionScoped lock(_netEqCritSect);
     for(WebRtc_Word16 idx = 0; idx < _numSlaves + 1; idx++)
     {
         if(!_isInitialized[idx])
@@ -1003,7 +1003,7 @@ ACMNetEQ::RemoveCodec(
             "RemoveCodec: NetEq error: could not Remove Codec, codec index out of range");
         return -1;
     }
-    CriticalSectionScoped lock(*_netEqCritSect);
+    CriticalSectionScoped lock(_netEqCritSect);
     if(!_isInitialized[0])
     {
         WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
@@ -1033,7 +1033,7 @@ WebRtc_Word16
 ACMNetEQ::SetBackgroundNoiseMode(
     const ACMBackgroundNoiseMode mode)
 {
-    CriticalSectionScoped lock(*_netEqCritSect);
+    CriticalSectionScoped lock(_netEqCritSect);
     for(WebRtc_Word16 idx = 0; idx < _numSlaves + 1; idx++)
     {
         if(!_isInitialized[idx])
@@ -1056,7 +1056,7 @@ ACMNetEQ::BackgroundNoiseMode(
     ACMBackgroundNoiseMode& mode)
 {
     WebRtcNetEQBGNMode myMode;
-    CriticalSectionScoped lock(*_netEqCritSect);
+    CriticalSectionScoped lock(_netEqCritSect);
     if(!_isInitialized[0])
     {
         WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
@@ -1079,7 +1079,7 @@ void
 ACMNetEQ::SetUniqueId(
     WebRtc_Word32 id)
 {
-    CriticalSectionScoped lock(*_netEqCritSect);
+    CriticalSectionScoped lock(_netEqCritSect);
     _id = id;
 }
 
@@ -1109,7 +1109,7 @@ WebRtc_Word32
 ACMNetEQ::PlayoutTimestamp(
     WebRtc_UWord32& timestamp)
 {
-    CriticalSectionScoped lock(*_netEqCritSect);
+    CriticalSectionScoped lock(_netEqCritSect);
     if(WebRtcNetEQ_GetSpeechTimeStamp(_inst[0], &timestamp) < 0)
     {
         LogError("GetSpeechTimeStamp", 0);
@@ -1122,7 +1122,7 @@ ACMNetEQ::PlayoutTimestamp(
 }
 
 void ACMNetEQ::RemoveSlaves() {
-  CriticalSectionScoped lock(*_netEqCritSect);
+  CriticalSectionScoped lock(_netEqCritSect);
   RemoveSlavesSafe();
 }
 
@@ -1158,7 +1158,7 @@ ACMNetEQ::AddSlave(
     const WebRtcNetEQDecoder* usedCodecs,
     WebRtc_Word16       noOfCodecs)
 {
-    CriticalSectionScoped lock(*_netEqCritSect);
+    CriticalSectionScoped lock(_netEqCritSect);
     const WebRtc_Word16 slaveIdx = 1;
     if(_numSlaves < 1)
     {
@@ -1263,14 +1263,14 @@ void
 ACMNetEQ::SetReceivedStereo(
     bool receivedStereo)
 {
-    CriticalSectionScoped lock(*_netEqCritSect);
+    CriticalSectionScoped lock(_netEqCritSect);
     _receivedStereo = receivedStereo;
 }
 
 WebRtc_UWord8
 ACMNetEQ::NumSlaves()
 {
-    CriticalSectionScoped lock(*_netEqCritSect);
+    CriticalSectionScoped lock(_netEqCritSect);
     return _numSlaves;
 }
 

src/modules/audio_coding/main/source/acm_resampler.cc

@@ -1,5 +1,5 @@
 /*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
@@ -11,73 +11,62 @@
 #include <string.h>
 
 #include "acm_resampler.h"
+
 #include "critical_section_wrapper.h"
 #include "resampler.h"
 #include "signal_processing_library.h"
 #include "trace.h"
 
-namespace webrtc
+namespace webrtc {
-{
 
-ACMResampler::ACMResampler():
+ACMResampler::ACMResampler()
-
+    : _resamplerCritSect(CriticalSectionWrapper::CreateCriticalSection()) {
-_resamplerCritSect(*CriticalSectionWrapper::CreateCriticalSection())
-{
 }
 
-ACMResampler::~ACMResampler()
+ACMResampler::~ACMResampler() {
-{
+  delete _resamplerCritSect;
-
-    delete &_resamplerCritSect;
 }
 
+WebRtc_Word16 ACMResampler::Resample10Msec(const WebRtc_Word16* inAudio,
+                                           WebRtc_Word32 inFreqHz,
+                                           WebRtc_Word16* outAudio,
+                                           WebRtc_Word32 outFreqHz,
+                                           WebRtc_UWord8 numAudioChannels) {
+  CriticalSectionScoped cs(_resamplerCritSect);
 
-WebRtc_Word16
+  if (inFreqHz == outFreqHz) {
-ACMResampler::Resample10Msec(
+    size_t length = static_cast<size_t>(inFreqHz * numAudioChannels / 100);
-    const WebRtc_Word16* inAudio,
+    memcpy(outAudio, inAudio, length * sizeof(WebRtc_Word16));
-    WebRtc_Word32        inFreqHz,
+    return static_cast<WebRtc_Word16>(inFreqHz / 100);
-    WebRtc_Word16*       outAudio,
+  }
-    WebRtc_Word32        outFreqHz,
-    WebRtc_UWord8        numAudioChannels)
-{
 
-    CriticalSectionScoped cs(_resamplerCritSect);
+  // |maxLen| is maximum number of samples for 10ms at 48kHz.
+  int maxLen = 480 * numAudioChannels;
+  int lengthIn = (WebRtc_Word16)(inFreqHz / 100) * numAudioChannels;
+  int outLen;
 
-    if(inFreqHz == outFreqHz)
+  WebRtc_Word32 ret;
-    {
+  ResamplerType type;
-        size_t length = static_cast<size_t>(inFreqHz * numAudioChannels / 100);
+  type = (numAudioChannels == 1) ? kResamplerSynchronous :
-        memcpy(outAudio, inAudio, length * sizeof(WebRtc_Word16));
+      kResamplerSynchronousStereo;
-        return static_cast<WebRtc_Word16>(inFreqHz / 100);
-    }
 
-    int maxLen = 480 * numAudioChannels; //max number of samples for 10ms at 48kHz
+  ret = _resampler.ResetIfNeeded(inFreqHz, outFreqHz, type);
-    int lengthIn = (WebRtc_Word16)(inFreqHz / 100) * numAudioChannels;
+  if (ret < 0) {
-    int outLen;
+    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, 0,
+                 "Error in reset of resampler");
+    return -1;
+  }
 
-    WebRtc_Word32 ret;
+  ret = _resampler.Push(inAudio, lengthIn, outAudio, maxLen, outLen);
-    ResamplerType type;
+  if (ret < 0) {
-    type = (numAudioChannels == 1)? kResamplerSynchronous:kResamplerSynchronousStereo;
+    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, 0,
+                 "Error in resampler: resampler.Push");
+    return -1;
+  }
 
-    ret = _resampler.ResetIfNeeded(inFreqHz,outFreqHz,type);
+  WebRtc_Word16 outAudioLenSmpl = (WebRtc_Word16) outLen / numAudioChannels;
-    if (ret < 0)
-    {
-        WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, 0,
-            "Error in reset of resampler");
-        return -1;
-    }
-
-    ret = _resampler.Push(inAudio, lengthIn, outAudio, maxLen, outLen);
-    if (ret < 0 )
-    {
-        WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, 0,
-            "Error in resampler: resampler.Push");
-        return -1;
-    }
-
-   WebRtc_Word16 outAudioLenSmpl = (WebRtc_Word16) outLen / numAudioChannels;
-
-   return outAudioLenSmpl;
 
+  return outAudioLenSmpl;
 }
 
-} // namespace webrtc
+}  // namespace webrtc

src/modules/audio_coding/main/source/acm_resampler.h

@@ -1,5 +1,5 @@
 /*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
@@ -18,26 +18,23 @@ namespace webrtc {
 
 class CriticalSectionWrapper;
 
-class ACMResampler
+class ACMResampler {
-{
+ public:
-public:
+  ACMResampler();
-    ACMResampler();
+  ~ACMResampler();
-    ~ACMResampler();
 
-    WebRtc_Word16 Resample10Msec(
+  WebRtc_Word16 Resample10Msec(const WebRtc_Word16* inAudio,
-        const WebRtc_Word16* inAudio,
+                               const WebRtc_Word32 inFreqHz,
-        const WebRtc_Word32  inFreqHz,
+                               WebRtc_Word16* outAudio,
-        WebRtc_Word16*       outAudio,
+                               const WebRtc_Word32 outFreqHz,
-        const WebRtc_Word32  outFreqHz,
+                               WebRtc_UWord8 numAudioChannels);
-        WebRtc_UWord8        numAudioChannels);
 
-private:
+ private:
-
+  // Use the Resampler class.
-    //Use the Resampler class
+  Resampler _resampler;
-    Resampler               _resampler;
+  CriticalSectionWrapper* _resamplerCritSect;
-    CriticalSectionWrapper& _resamplerCritSect;
 };
 
-} // namespace webrtc
+}  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_RESAMPLER_H_

src/modules/audio_coding/main/source/audio_coding_module_impl.cc

@@ -137,7 +137,7 @@ AudioCodingModuleImpl::AudioCodingModuleImpl(const WebRtc_Word32 id)
 
 AudioCodingModuleImpl::~AudioCodingModuleImpl() {
   {
-    CriticalSectionScoped lock(*_acmCritSect);
+    CriticalSectionScoped lock(_acmCritSect);
     _currentSendCodecIdx = -1;
 
     for (int i = 0; i < ACMCodecDB::kMaxNumCodecs; i++) {
@@ -211,7 +211,7 @@ AudioCodingModuleImpl::~AudioCodingModuleImpl() {
 
 WebRtc_Word32 AudioCodingModuleImpl::ChangeUniqueId(const WebRtc_Word32 id) {
   {
-    CriticalSectionScoped lock(*_acmCritSect);
+    CriticalSectionScoped lock(_acmCritSect);
     _id = id;
 
 #ifdef ACM_QA_TEST
@@ -246,7 +246,7 @@ WebRtc_Word32 AudioCodingModuleImpl::ChangeUniqueId(const WebRtc_Word32 id) {
 // Returns the number of milliseconds until the module want a
 // worker thread to call Process.
 WebRtc_Word32 AudioCodingModuleImpl::TimeUntilNextProcess() {
-  CriticalSectionScoped lock(*_acmCritSect);
+  CriticalSectionScoped lock(_acmCritSect);
 
   if (!HaveValidEncoder("TimeUntilNextProcess")) {
     return -1;
@@ -271,7 +271,7 @@ WebRtc_Word32 AudioCodingModuleImpl::Process() {
 
   // Keep the scope of the ACM critical section limited.
   {
-    CriticalSectionScoped lock(*_acmCritSect);
+    CriticalSectionScoped lock(_acmCritSect);
     if (!HaveValidEncoder("Process")) {
       return -1;
     }
@@ -428,7 +428,7 @@ WebRtc_Word32 AudioCodingModuleImpl::Process() {
   }
 
   if (has_data_to_send) {
-    CriticalSectionScoped lock(*_callbackCritSect);
+    CriticalSectionScoped lock(_callbackCritSect);
 #ifdef ACM_QA_TEST
     if(_outgoingPL != NULL) {
       if (fwrite(&rtp_timestamp, sizeof(WebRtc_UWord32), 1, _outgoingPL) != 1) {
@@ -476,7 +476,7 @@ WebRtc_Word32 AudioCodingModuleImpl::Process() {
 
 // Initialize send codec.
 WebRtc_Word32 AudioCodingModuleImpl::InitializeSender() {
-  CriticalSectionScoped lock(*_acmCritSect);
+  CriticalSectionScoped lock(_acmCritSect);
 
   // Start with invalid values.
   _sendCodecRegistered = false;
@@ -512,7 +512,7 @@ WebRtc_Word32 AudioCodingModuleImpl::InitializeSender() {
 }
 
 WebRtc_Word32 AudioCodingModuleImpl::ResetEncoder() {
-  CriticalSectionScoped lock(*_acmCritSect);
+  CriticalSectionScoped lock(_acmCritSect);
   if (!HaveValidEncoder("ResetEncoder")) {
     return -1;
   }
@@ -520,7 +520,7 @@ WebRtc_Word32 AudioCodingModuleImpl::ResetEncoder() {
 }
 
 void AudioCodingModuleImpl::UnregisterSendCodec() {
-  CriticalSectionScoped lock(*_acmCritSect);
+  CriticalSectionScoped lock(_acmCritSect);
   _sendCodecRegistered = false;
   _currentSendCodecIdx = -1;
   return;
@@ -557,7 +557,7 @@ WebRtc_Word32 AudioCodingModuleImpl::RegisterSendCodec(
   int mirror_id;
   int codec_id = ACMCodecDB::CodecNumber(&send_codec, &mirror_id, error_message,
                                         sizeof(error_message));
-  CriticalSectionScoped lock(*_acmCritSect);
+  CriticalSectionScoped lock(_acmCritSect);
 
   // Check for reported errors from function CodecNumber().
   if (codec_id < 0) {
@@ -815,7 +815,7 @@ WebRtc_Word32 AudioCodingModuleImpl::SendCodec(
     CodecInst& current_codec) const {
   WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, _id,
                "SendCodec()");
-  CriticalSectionScoped lock(*_acmCritSect);
+  CriticalSectionScoped lock(_acmCritSect);
 
   if (!_sendCodecRegistered) {
     WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, _id,
@@ -835,7 +835,7 @@ WebRtc_Word32 AudioCodingModuleImpl::SendCodec(
 WebRtc_Word32 AudioCodingModuleImpl::SendFrequency() const {
   WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, _id,
                "SendFrequency()");
-  CriticalSectionScoped lock(*_acmCritSect);
+  CriticalSectionScoped lock(_acmCritSect);
 
   if (!_sendCodecRegistered) {
     WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, _id,
@@ -851,7 +851,7 @@ WebRtc_Word32 AudioCodingModuleImpl::SendFrequency() const {
 // Adaptive rate codecs return their current encode target rate, while other
 // codecs return there longterm avarage or their fixed rate.
 WebRtc_Word32 AudioCodingModuleImpl::SendBitrate() const {
-  CriticalSectionScoped lock(*_acmCritSect);
+  CriticalSectionScoped lock(_acmCritSect);
 
   if (!_sendCodecRegistered) {
     WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, _id,
@@ -877,7 +877,7 @@ WebRtc_Word32 AudioCodingModuleImpl::SetReceivedEstimatedBandwidth(
 // the encoded buffers.
 WebRtc_Word32 AudioCodingModuleImpl::RegisterTransportCallback(
     AudioPacketizationCallback* transport) {
-  CriticalSectionScoped lock(*_callbackCritSect);
+  CriticalSectionScoped lock(_callbackCritSect);
   _packetizationCallback = transport;
   return 0;
 }
@@ -896,12 +896,12 @@ WebRtc_Word32 AudioCodingModuleImpl::RegisterIncomingMessagesCallback(
 
   // Enter the critical section for callback.
   {
-    CriticalSectionScoped lock(*_callbackCritSect);
+    CriticalSectionScoped lock(_callbackCritSect);
     _dtmfCallback = incoming_message;
   }
   // Enter the ACM critical section to set up the DTMF class.
   {
-    CriticalSectionScoped lock(*_acmCritSect);
+    CriticalSectionScoped lock(_acmCritSect);
     // Check if the call is to disable or enable the callback.
     if (incoming_message == NULL) {
       // Callback is disabled, delete DTMF-detector class.
@@ -932,7 +932,7 @@ WebRtc_Word32 AudioCodingModuleImpl::RegisterIncomingMessagesCallback(
   // Check if we failed in setting up the DTMF-detector class.
   if ((status < 0)) {
     // We failed, we cannot have the callback.
-    CriticalSectionScoped lock(*_callbackCritSect);
+    CriticalSectionScoped lock(_callbackCritSect);
     _dtmfCallback = NULL;
   }
 
@@ -944,7 +944,7 @@ WebRtc_Word32 AudioCodingModuleImpl::RegisterIncomingMessagesCallback(
 WebRtc_Word32 AudioCodingModuleImpl::Add10MsData(
     const AudioFrame& audio_frame) {
   // Do we have a codec registered?
-  CriticalSectionScoped lock(*_acmCritSect);
+  CriticalSectionScoped lock(_acmCritSect);
   if (!HaveValidEncoder("Add10MsData")) {
     return -1;
   }
@@ -1058,7 +1058,7 @@ WebRtc_Word32 AudioCodingModuleImpl::Add10MsData(
 //
 
 bool AudioCodingModuleImpl::FECStatus() const {
-  CriticalSectionScoped lock(*_acmCritSect);
+  CriticalSectionScoped lock(_acmCritSect);
   return _fecEnabled;
 }
 
@@ -1067,7 +1067,7 @@ WebRtc_Word32
 AudioCodingModuleImpl::SetFECStatus(
 #ifdef WEBRTC_CODEC_RED
     const bool enable_fec) {
-  CriticalSectionScoped lock(*_acmCritSect);
+  CriticalSectionScoped lock(_acmCritSect);
 
   if (_fecEnabled != enable_fec) {
     // Reset the RED buffer.
@@ -1104,7 +1104,7 @@ AudioCodingModuleImpl::SetFECStatus(
 WebRtc_Word32 AudioCodingModuleImpl::SetVAD(const bool enable_dtx,
                                             const bool enable_vad,
                                             const ACMVADMode mode) {
-  CriticalSectionScoped lock(*_acmCritSect);
+  CriticalSectionScoped lock(_acmCritSect);
 
   // Sanity check of the mode.
   if ((mode != VADNormal) && (mode != VADLowBitrate)
@@ -1149,7 +1149,7 @@ WebRtc_Word32 AudioCodingModuleImpl::SetVAD(const bool enable_dtx,
 // Get VAD/DTX settings.
 WebRtc_Word32 AudioCodingModuleImpl::VAD(bool& dtx_enabled, bool& vad_enabled,
                                          ACMVADMode& mode) const {
-  CriticalSectionScoped lock(*_acmCritSect);
+  CriticalSectionScoped lock(_acmCritSect);
 
   dtx_enabled = _dtxEnabled;
   vad_enabled = _vadEnabled;
@@ -1163,7 +1163,7 @@ WebRtc_Word32 AudioCodingModuleImpl::VAD(bool& dtx_enabled, bool& vad_enabled,
 //
 
 WebRtc_Word32 AudioCodingModuleImpl::InitializeReceiver() {
-  CriticalSectionScoped lock(*_acmCritSect);
+  CriticalSectionScoped lock(_acmCritSect);
   return InitializeReceiverSafe();
 }
 
@@ -1213,7 +1213,7 @@ WebRtc_Word32 AudioCodingModuleImpl::InitializeReceiverSafe() {
 
 // Reset the decoder state.
 WebRtc_Word32 AudioCodingModuleImpl::ResetDecoder() {
-  CriticalSectionScoped lock(*_acmCritSect);
+  CriticalSectionScoped lock(_acmCritSect);
 
   for (int id = 0; id < ACMCodecDB::kMaxNumCodecs; id++) {
     if ((_codecs[id] != NULL) && (_registeredPlTypes[id] != -1)) {
@@ -1233,7 +1233,7 @@ WebRtc_Word32 AudioCodingModuleImpl::ReceiveFrequency() const {
                "ReceiveFrequency()");
   WebRtcACMCodecParams codec_params;
 
-  CriticalSectionScoped lock(*_acmCritSect);
+  CriticalSectionScoped lock(_acmCritSect);
   if (DecoderParamByPlType(_lastRecvAudioCodecPlType, codec_params) < 0) {
     return _netEq.CurrentSampFreqHz();
   } else {
@@ -1246,7 +1246,7 @@ WebRtc_Word32 AudioCodingModuleImpl::PlayoutFrequency() const {
   WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, _id,
                "PlayoutFrequency()");
 
-  CriticalSectionScoped lock(*_acmCritSect);
+  CriticalSectionScoped lock(_acmCritSect);
 
   return _netEq.CurrentSampFreqHz();
 }
@@ -1255,7 +1255,7 @@ WebRtc_Word32 AudioCodingModuleImpl::PlayoutFrequency() const {
 // for codecs, CNG (NB, WB and SWB), DTMF, RED.
 WebRtc_Word32 AudioCodingModuleImpl::RegisterReceiveCodec(
     const CodecInst& receive_codec) {
-  CriticalSectionScoped lock(*_acmCritSect);
+  CriticalSectionScoped lock(_acmCritSect);
 
   if (receive_codec.channels > 2) {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
@@ -1456,7 +1456,7 @@ WebRtc_Word32 AudioCodingModuleImpl::RegisterRecCodecMSSafe(
 WebRtc_Word32 AudioCodingModuleImpl::ReceiveCodec(
     CodecInst& current_codec) const {
   WebRtcACMCodecParams decoderParam;
-  CriticalSectionScoped lock(*_acmCritSect);
+  CriticalSectionScoped lock(_acmCritSect);
 
   for (int id = 0; id < ACMCodecDB::kMaxNumCodecs; id++) {
     if (_codecs[id] != NULL) {
@@ -1493,7 +1493,7 @@ WebRtc_Word32 AudioCodingModuleImpl::IncomingPacket(
   {
     // Store the payload Type. This will be used to retrieve "received codec"
     // and "received frequency."
-    CriticalSectionScoped lock(*_acmCritSect);
+    CriticalSectionScoped lock(_acmCritSect);
 #ifdef ACM_QA_TEST
     if(_incomingPL != NULL) {
       if (fwrite(&rtp_info.header.timestamp, sizeof(WebRtc_UWord32),
@@ -1763,7 +1763,7 @@ WebRtc_Word32 AudioCodingModuleImpl::PlayoutData10Ms(
 
   // Limit the scope of ACM Critical section.
   {
-    CriticalSectionScoped lock(*_acmCritSect);
+    CriticalSectionScoped lock(_acmCritSect);
 
     if ((receive_freq != desired_freq_hz) && (desired_freq_hz != -1)) {
       // Resample payloadData.
@@ -1844,7 +1844,7 @@ WebRtc_Word32 AudioCodingModuleImpl::PlayoutData10Ms(
 
   if (tone_detected) {
     // We will deal with callback here, so enter callback critical section.
-    CriticalSectionScoped lock(*_callbackCritSect);
+    CriticalSectionScoped lock(_callbackCritSect);
 
     if (_dtmfCallback != NULL) {
       if (tone != kACMToneEnd) {
@@ -1936,7 +1936,7 @@ WebRtc_Word32 AudioCodingModuleImpl::RegisterVADCallback(
     ACMVADCallback* vad_callback) {
   WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceAudioCoding, _id,
                "RegisterVADCallback()");
-  CriticalSectionScoped lock(*_callbackCritSect);
+  CriticalSectionScoped lock(_callbackCritSect);
   _vadCallback = vad_callback;
   return 0;
 }
@@ -2017,7 +2017,7 @@ WebRtc_Word32 AudioCodingModuleImpl::IncomingPayload(
 WebRtc_Word16 AudioCodingModuleImpl::DecoderParamByPlType(
     const WebRtc_UWord8 payload_type,
     WebRtcACMCodecParams& codec_params) const {
-  CriticalSectionScoped lock(*_acmCritSect);
+  CriticalSectionScoped lock(_acmCritSect);
   for (WebRtc_Word16 id = 0; id < ACMCodecDB::kMaxNumCodecs;
       id++) {
     if (_codecs[id] != NULL) {
@@ -2041,7 +2041,7 @@ WebRtc_Word16 AudioCodingModuleImpl::DecoderParamByPlType(
 WebRtc_Word16 AudioCodingModuleImpl::DecoderListIDByPlName(
     const char* name, const WebRtc_UWord16 frequency) const {
   WebRtcACMCodecParams codec_params;
-  CriticalSectionScoped lock(*_acmCritSect);
+  CriticalSectionScoped lock(_acmCritSect);
   for (WebRtc_Word16 id = 0; id < ACMCodecDB::kMaxNumCodecs; id++) {
     if ((_codecs[id] != NULL)) {
       if (_codecs[id]->DecoderInitialized()) {
@@ -2071,7 +2071,7 @@ WebRtc_Word16 AudioCodingModuleImpl::DecoderListIDByPlName(
 
 WebRtc_Word32 AudioCodingModuleImpl::LastEncodedTimestamp(
     WebRtc_UWord32& timestamp) const {
-  CriticalSectionScoped lock(*_acmCritSect);
+  CriticalSectionScoped lock(_acmCritSect);
   if (!HaveValidEncoder("LastEncodedTimestamp")) {
     return -1;
   }
@@ -2081,7 +2081,7 @@ WebRtc_Word32 AudioCodingModuleImpl::LastEncodedTimestamp(
 
 WebRtc_Word32 AudioCodingModuleImpl::ReplaceInternalDTXWithWebRtc(
     bool use_webrtc_dtx) {
-  CriticalSectionScoped lock(*_acmCritSect);
+  CriticalSectionScoped lock(_acmCritSect);
 
   if (!HaveValidEncoder("ReplaceInternalDTXWithWebRtc")) {
     WEBRTC_TRACE(
@@ -2107,7 +2107,7 @@ WebRtc_Word32 AudioCodingModuleImpl::ReplaceInternalDTXWithWebRtc(
 
 WebRtc_Word32 AudioCodingModuleImpl::IsInternalDTXReplacedWithWebRtc(
     bool& uses_webrtc_dtx) {
-  CriticalSectionScoped lock(*_acmCritSect);
+  CriticalSectionScoped lock(_acmCritSect);
 
   if (!HaveValidEncoder("IsInternalDTXReplacedWithWebRtc")) {
     return -1;
@@ -2121,7 +2121,7 @@ WebRtc_Word32 AudioCodingModuleImpl::IsInternalDTXReplacedWithWebRtc(
 
 WebRtc_Word32 AudioCodingModuleImpl::SetISACMaxRate(
     const WebRtc_UWord32 max_bit_per_sec) {
-  CriticalSectionScoped lock(*_acmCritSect);
+  CriticalSectionScoped lock(_acmCritSect);
 
   if (!HaveValidEncoder("SetISACMaxRate")) {
     return -1;
@@ -2132,7 +2132,7 @@ WebRtc_Word32 AudioCodingModuleImpl::SetISACMaxRate(
 
 WebRtc_Word32 AudioCodingModuleImpl::SetISACMaxPayloadSize(
     const WebRtc_UWord16 max_size_bytes) {
-  CriticalSectionScoped lock(*_acmCritSect);
+  CriticalSectionScoped lock(_acmCritSect);
 
   if (!HaveValidEncoder("SetISACMaxPayloadSize")) {
     return -1;
@@ -2146,7 +2146,7 @@ WebRtc_Word32 AudioCodingModuleImpl::ConfigISACBandwidthEstimator(
     const WebRtc_UWord8 frame_size_ms,
     const WebRtc_UWord16 rate_bit_per_sec,
     const bool enforce_frame_size) {
-  CriticalSectionScoped lock(*_acmCritSect);
+  CriticalSectionScoped lock(_acmCritSect);
 
   if (!HaveValidEncoder("ConfigISACBandwidthEstimator")) {
     return -1;
@@ -2201,7 +2201,7 @@ bool AudioCodingModuleImpl::HaveValidEncoder(const char* caller_name) const {
 
 WebRtc_Word32 AudioCodingModuleImpl::UnregisterReceiveCodec(
     const WebRtc_Word16 payload_type) {
-  CriticalSectionScoped lock(*_acmCritSect);
+  CriticalSectionScoped lock(_acmCritSect);
   int id;
 
   // Search through the list of registered payload types.