Switch CriticalSectionWrapper->rtc::CriticalSection in modules/audio_coding.

webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h

 /*
  *  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
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_AUDIO_CODING_MODULE_IMPL_H_
 #define WEBRTC_MODULES_AUDIO_CODING_ACM2_AUDIO_CODING_MODULE_IMPL_H_
 
 #include <string>
 #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"
 #include "webrtc/engine_configurations.h"
 #include "webrtc/modules/audio_coding/acm2/acm_receiver.h"
 #include "webrtc/modules/audio_coding/acm2/acm_resampler.h"
 #include "webrtc/modules/audio_coding/acm2/codec_manager.h"
 
 namespace webrtc {
 
-class CriticalSectionWrapper;
 class AudioCodingImpl;
 
 namespace acm2 {
 
 class AudioCodingModuleImpl final : public AudioCodingModule {
  public:
   friend webrtc::AudioCodingImpl;
 
   explicit AudioCodingModuleImpl(const AudioCodingModule::Config& config);
   ~AudioCodingModuleImpl() override;
@@ skipping 194 matching lines @@
   //   -1: if encountering an error.
   //    0: otherwise.
   int PreprocessToAddData(const AudioFrame& in_frame,
                           const AudioFrame** ptr_out)
       EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
 
   // Change required states after starting to receive the codec corresponding
   // 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_);
   uint32_t expected_in_ts_ GUARDED_BY(acm_crit_sect_);
   ACMResampler resampler_ GUARDED_BY(acm_crit_sect_);
   AcmReceiver receiver_;  // AcmReceiver has it's own internal lock.
   ChangeLogger bitrate_logger_ GUARDED_BY(acm_crit_sect_);
   CodecManager codec_manager_ GUARDED_BY(acm_crit_sect_);
   RentACodec rent_a_codec_ GUARDED_BY(acm_crit_sect_);
 
@@ skipping 10 matching lines @@
 
   bool receiver_initialized_ GUARDED_BY(acm_crit_sect_);
 
   AudioFrame preprocess_frame_ GUARDED_BY(acm_crit_sect_);
   bool first_10ms_data_ GUARDED_BY(acm_crit_sect_);
 
   bool first_frame_ GUARDED_BY(acm_crit_sect_);
   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_);
 };
 
 }  // namespace acm2
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_AUDIO_CODING_MODULE_IMPL_H_