Get rid of the manual destructor in AudioCodingModuleImpl

webrtc/modules/audio_coding/main/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.
  */
 
@@ skipping 246 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);
 
-  CriticalSectionWrapper* acm_crit_sect_;
+  const rtc::scoped_ptr<CriticalSectionWrapper> 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_);
 
   // This is to keep track of CN instances where we can send DTMFs.
   uint8_t previous_pltype_ GUARDED_BY(acm_crit_sect_);
 
   // Used when payloads are pushed into ACM without any RTP info
   // One example is when pre-encoded bit-stream is pushed from
   // a file.
   // IMPORTANT: this variable is only used in IncomingPayload(), therefore,
   // no lock acquired when interacting with this variable. If it is going to
   // be used in other methods, locks need to be taken.
-  WebRtcRTPHeader* aux_rtp_header_;
+  rtc::scoped_ptr<WebRtcRTPHeader> aux_rtp_header_;
 
   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_);
 
-  CriticalSectionWrapper* callback_crit_sect_;
+  const rtc::scoped_ptr<CriticalSectionWrapper> callback_crit_sect_;
   AudioPacketizationCallback* packetization_callback_
       GUARDED_BY(callback_crit_sect_);
   ACMVADCallback* vad_callback_ GUARDED_BY(callback_crit_sect_);
 };
 
 }  // namespace acm2
 
 class AudioCodingImpl : public AudioCoding {
  public:
   AudioCodingImpl(const Config& config);
@@ skipping 62 matching lines @@
   int playout_frequency_hz_;
   // TODO(henrik.lundin): All members below this line are temporary and should
   // be removed after refactoring is completed.
   rtc::scoped_ptr<acm2::AudioCodingModuleImpl> acm_old_;
   CodecInst current_send_codec_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_AUDIO_CODING_MODULE_IMPL_H_