Remove mutable from rtc::CriticalSections.

webrtc/voice_engine/channel.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 134 matching lines @@
     rtc::CritScope lock(&lock_);
     state_.sending = enable;
   }
 
   void SetReceiving(bool enable) {
     rtc::CritScope lock(&lock_);
     state_.receiving = enable;
   }
 
  private:
-  mutable rtc::CriticalSection lock_;
+  rtc::CriticalSection lock_;
   State state_;
 };
 
 class Channel
     : public RtpData,
       public RtpFeedback,
       public FileCallback,  // receiving notification from file player &
                             // recorder
       public Transport,
       public RtpAudioFeedback,
@@ skipping 310 matching lines @@
   void RegisterReceiveCodecsToRTPModule();
 
   int SetRedPayloadType(int red_payload_type);
   int SetSendRtpHeaderExtension(bool enable,
                                 RTPExtensionType type,
                                 unsigned char id);
 
   int32_t GetPlayoutFrequency();
   int64_t GetRTT(bool allow_associate_channel) const;
 
-  mutable rtc::CriticalSection _fileCritSect;
-  mutable rtc::CriticalSection _callbackCritSect;
-  mutable rtc::CriticalSection volume_settings_critsect_;
+  rtc::CriticalSection _fileCritSect;
+  rtc::CriticalSection _callbackCritSect;
+  rtc::CriticalSection volume_settings_critsect_;
   uint32_t _instanceId;
   int32_t _channelId;
 
   ChannelState channel_state_;
 
   RtcEventLog* const event_log_;
 
   rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_;
   rtc::scoped_ptr<RTPPayloadRegistry> rtp_payload_registry_;
   rtc::scoped_ptr<ReceiveStatistics> rtp_receive_statistics_;
@@ skipping 27 matching lines @@
 
   // Timestamp of the audio pulled from NetEq.
   uint32_t jitter_buffer_playout_timestamp_;
   uint32_t playout_timestamp_rtp_ GUARDED_BY(video_sync_lock_);
   uint32_t playout_timestamp_rtcp_;
   uint32_t playout_delay_ms_ GUARDED_BY(video_sync_lock_);
   uint32_t _numberOfDiscardedPackets;
   uint16_t send_sequence_number_;
   uint8_t restored_packet_[kVoiceEngineMaxIpPacketSizeBytes];
 
-  mutable rtc::CriticalSection ts_stats_lock_;
+  rtc::CriticalSection ts_stats_lock_;
 
   rtc::scoped_ptr<rtc::TimestampWrapAroundHandler> rtp_ts_wraparound_handler_;
   // The rtp timestamp of the first played out audio frame.
   int64_t capture_start_rtp_time_stamp_;
   // The capture ntp time (in local timebase) of the first played out audio
   // frame.
   int64_t capture_start_ntp_time_ms_ GUARDED_BY(ts_stats_lock_);
 
   // uses
   Statistics* _engineStatisticsPtr;
@@ skipping 20 matching lines @@
   // VoEDtmf
   bool _playOutbandDtmfEvent;
   bool _playInbandDtmfEvent;
   // VoeRTP_RTCP
   uint32_t _lastLocalTimeStamp;
   int8_t _lastPayloadType;
   bool _includeAudioLevelIndication;
   // VoENetwork
   AudioFrame::SpeechType _outputSpeechType;
   // VoEVideoSync
-  mutable rtc::CriticalSection video_sync_lock_;
+  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_);
   // VoEAudioProcessing
   bool _RxVadDetection;
   bool _rxAgcIsEnabled;
   bool _rxNsIsEnabled;
   bool restored_packet_in_use_;
   // RtcpBandwidthObserver
   rtc::scoped_ptr<VoERtcpObserver> rtcp_observer_;
   rtc::scoped_ptr<NetworkPredictor> network_predictor_;
   // An associated send channel.
-  mutable rtc::CriticalSection assoc_send_channel_lock_;
+  rtc::CriticalSection assoc_send_channel_lock_;
   ChannelOwner associate_send_channel_ GUARDED_BY(assoc_send_channel_lock_);
 
   bool pacing_enabled_;
   PacketRouter* packet_router_ = nullptr;
   rtc::scoped_ptr<TransportFeedbackProxy> feedback_observer_proxy_;
   rtc::scoped_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_;
   rtc::scoped_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_;
 };
 
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_CHANNEL_H_