Remove mutable from rtc::CriticalSections.

A couple of mutables were added after last removal of mutables, so
removing those. rtc::CriticalSection is const-lockable.

BUG=
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1652983002

Cr-Commit-Position: refs/heads/master@{#11447}

webrtc/video/call_stats.h

@@ -60,7 +60,7 @@ class CallStats : public Module {
  private:
   Clock* const clock_;
   // Protecting all members.
-  mutable rtc::CriticalSection crit_;
+  rtc::CriticalSection crit_;
   // Observer receiving statistics updates.
   rtc::scoped_ptr<RtcpRttStats> rtcp_rtt_stats_;
   // The last time 'Process' resulted in statistic update.

webrtc/video/encoder_state_feedback.h

@@ -55,7 +55,7 @@ class EncoderStateFeedback {
  private:
   typedef std::map<uint32_t,  ViEEncoder*> SsrcEncoderMap;
 
-  mutable rtc::CriticalSection crit_;
+  rtc::CriticalSection crit_;
 
   // Instance registered at the class requesting new key frames.
   rtc::scoped_ptr<EncoderStateFeedbackObserver> observer_;

webrtc/video/payload_router.h

@@ -69,7 +69,7 @@ class PayloadRouter {
  private:
   // TODO(mflodman): When the new video API has launched, remove crit_ and
   // assume rtp_modules_ will never change during a call.
-  mutable rtc::CriticalSection crit_;
+  rtc::CriticalSection crit_;
 
   // Active sending RTP modules, in layer order.
   std::vector<RtpRtcp*> rtp_modules_ GUARDED_BY(crit_);

webrtc/video/video_capture_input.h

@@ -63,7 +63,7 @@ class VideoCaptureInput : public webrtc::VideoCaptureInput {
   static bool EncoderThreadFunction(void* obj);
   bool EncoderProcess();
 
-  mutable rtc::CriticalSection crit_;
+  rtc::CriticalSection crit_;
   ProcessThread* const module_process_thread_;
 
   VideoCaptureCallback* const frame_callback_;

webrtc/video/vie_channel.h

@@ -332,7 +332,7 @@ class ViEChannel : public VCMFrameTypeCallback,
     // Note: this should be implemented with a RW-lock to allow simultaneous
     // calls into the callback. However that doesn't seem to be needed for the
     // current type of callbacks covered by this class.
-    mutable rtc::CriticalSection critsect_;
+    rtc::CriticalSection critsect_;
     T* callback_ GUARDED_BY(critsect_);
 
    private:
@@ -403,7 +403,7 @@ class ViEChannel : public VCMFrameTypeCallback,
   ProcessThread* const module_process_thread_;
 
   // Used for all registered callbacks except rendering.
-  mutable rtc::CriticalSection crit_;
+  rtc::CriticalSection crit_;
 
   // Owned modules/classes.
   rtc::scoped_refptr<PayloadRouter> send_payload_router_;

webrtc/video/vie_encoder.h

@@ -158,7 +158,7 @@ class ViEEncoder : public RtcpIntraFrameObserver,
   const rtc::scoped_ptr<VideoCodingModule> vcm_;
   rtc::scoped_refptr<PayloadRouter> send_payload_router_;
 
-  mutable rtc::CriticalSection data_cs_;
+  rtc::CriticalSection data_cs_;
   rtc::scoped_ptr<BitrateObserver> bitrate_observer_;
 
   SendStatisticsProxy* const stats_proxy_;

webrtc/video/vie_remb.h

@@ -58,7 +58,7 @@ class VieRemb : public RemoteBitrateObserver {
   typedef std::list<RtpRtcp*> RtpModules;
 
   Clock* const clock_;
-  mutable rtc::CriticalSection list_crit_;
+  rtc::CriticalSection list_crit_;
 
   // The last time a REMB was sent.
   int64_t last_remb_time_;

webrtc/voice_engine/channel.h

@@ -152,7 +152,7 @@ class ChannelState {
   }
 
  private:
-  mutable rtc::CriticalSection lock_;
+  rtc::CriticalSection lock_;
   State state_;
 };
 
@@ -483,9 +483,9 @@ class Channel
   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;
 
@@ -533,7 +533,7 @@ class Channel
   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.
@@ -574,7 +574,7 @@ class Channel
   // 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_);
@@ -587,7 +587,7 @@ class Channel
   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_;

webrtc/voice_engine/channel_manager.h

@@ -123,7 +123,7 @@ class ChannelManager {
 
   Atomic32 last_channel_id_;
 
-  mutable rtc::CriticalSection lock_;
+  rtc::CriticalSection lock_;
   std::vector<ChannelOwner> channels_;
 
   const Config& config_;

webrtc/voice_engine/dtmf_inband.h

@@ -67,7 +67,7 @@ private:
                                    int16_t length);
 
 private:
-    mutable rtc::CriticalSection _critSect;
+    rtc::CriticalSection _critSect;
     int32_t _id;
     uint16_t _outputFrequencyHz;  // {8000, 16000, 32000}
     int16_t _oldOutputLow[2];     // Data needed for oscillator model

webrtc/voice_engine/level_indicator.h

@@ -40,7 +40,7 @@ public:
 private:
     enum { kUpdateFrequency = 10};
 
-    mutable rtc::CriticalSection _critSect;
+    rtc::CriticalSection _critSect;
 
     int16_t _absMax;
     int16_t _count;

webrtc/voice_engine/shared_data.h

@@ -63,7 +63,7 @@ public:
 
 protected:
     const uint32_t _instanceId;
-    mutable rtc::CriticalSection _apiCritPtr;
+    rtc::CriticalSection _apiCritPtr;
     ChannelManager _channelManager;
     Statistics _engineStatistics;
     AudioDeviceModule* _audioDevicePtr;

webrtc/voice_engine/statistics.h

@@ -39,7 +39,7 @@ class Statistics
     int32_t LastError() const;
 
  private:
-    mutable rtc::CriticalSection lock_;
+    rtc::CriticalSection lock_;
     const uint32_t _instanceId;
     mutable int32_t _lastError;
     bool _isInitialized;

webrtc/voice_engine/test/auto_test/fakes/conference_transport.h

@@ -128,8 +128,8 @@ class ConferenceTransport: public webrtc::Transport {
   void SendPacket(const Packet& packet);
   bool DispatchPackets();
 
-  mutable rtc::CriticalSection pq_crit_;
-  mutable rtc::CriticalSection stream_crit_;
+  rtc::CriticalSection pq_crit_;
+  rtc::CriticalSection stream_crit_;
   const rtc::scoped_ptr<webrtc::EventWrapper> packet_event_;
   rtc::PlatformThread thread_;
 

webrtc/voice_engine/test/auto_test/fixtures/after_initialization_fixture.h

@@ -142,7 +142,7 @@ class LoopBackTransport : public webrtc::Transport {
     return true;
   }
 
-  mutable rtc::CriticalSection crit_;
+  rtc::CriticalSection crit_;
   const rtc::scoped_ptr<webrtc::EventWrapper> packet_event_;
   rtc::PlatformThread thread_;
   std::deque<Packet> packet_queue_ GUARDED_BY(crit_);