| Index: webrtc/voice_engine/channel.h
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| diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h
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| index 3ef5b4a2942fb2e0157f9fc8bf557c7273d019ad..3a4383ffd1e48003cb2144de8d10567c2d2cda01 100644
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| --- a/webrtc/voice_engine/channel.h
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| +++ b/webrtc/voice_engine/channel.h
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| @@ -47,7 +47,6 @@ namespace webrtc {
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|  
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|  class AudioDeviceModule;
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|  class Config;
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| -class CriticalSectionWrapper;
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|  class FileWrapper;
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|  class PacketRouter;
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|  class ProcessThread;
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| @@ -103,57 +102,56 @@ class ChannelState {
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|          bool receiving;
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|      };
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|  
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| -    ChannelState() : lock_(CriticalSectionWrapper::CreateCriticalSection()) {
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| -    }
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| +    ChannelState() {}
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|      virtual ~ChannelState() {}
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|  
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|      void Reset() {
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| -        CriticalSectionScoped lock(lock_.get());
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| +        rtc::CritScope lock(&lock_);
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|          state_ = State();
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|      }
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|  
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|      State Get() const {
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| -        CriticalSectionScoped lock(lock_.get());
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| +        rtc::CritScope lock(&lock_);
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|          return state_;
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|      }
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|  
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|      void SetRxApmIsEnabled(bool enable) {
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| -        CriticalSectionScoped lock(lock_.get());
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| +        rtc::CritScope lock(&lock_);
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|          state_.rx_apm_is_enabled = enable;
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|      }
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|  
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|      void SetInputExternalMedia(bool enable) {
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| -        CriticalSectionScoped lock(lock_.get());
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| +        rtc::CritScope lock(&lock_);
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|          state_.input_external_media = enable;
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|      }
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|  
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|      void SetOutputFilePlaying(bool enable) {
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| -        CriticalSectionScoped lock(lock_.get());
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| +        rtc::CritScope lock(&lock_);
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|          state_.output_file_playing = enable;
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|      }
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|  
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|      void SetInputFilePlaying(bool enable) {
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| -        CriticalSectionScoped lock(lock_.get());
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| +        rtc::CritScope lock(&lock_);
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|          state_.input_file_playing = enable;
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|      }
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|  
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|      void SetPlaying(bool enable) {
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| -        CriticalSectionScoped lock(lock_.get());
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| +        rtc::CritScope lock(&lock_);
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|          state_.playing = enable;
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|      }
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|  
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|      void SetSending(bool enable) {
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| -        CriticalSectionScoped lock(lock_.get());
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| +        rtc::CritScope lock(&lock_);
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|          state_.sending = enable;
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|      }
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|  
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|      void SetReceiving(bool enable) {
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| -        CriticalSectionScoped lock(lock_.get());
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| +        rtc::CritScope lock(&lock_);
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|          state_.receiving = enable;
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|      }
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|  
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|  private:
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| - rtc::scoped_ptr<CriticalSectionWrapper> lock_;
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| +    mutable rtc::CriticalSection lock_;
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|      State state_;
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|  };
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|  
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| @@ -190,7 +188,7 @@ public:
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|          ProcessThread& moduleProcessThread,
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|          AudioDeviceModule& audioDeviceModule,
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|          VoiceEngineObserver* voiceEngineObserver,
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| -        CriticalSectionWrapper* callbackCritSect);
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| +        rtc::CriticalSection* callbackCritSect);
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|      int32_t UpdateLocalTimeStamp();
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|  
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|      void SetSink(rtc::scoped_ptr<AudioSinkInterface> sink);
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| @@ -430,7 +428,7 @@ public:
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|      }
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|      bool ExternalTransport() const
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|      {
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| -        CriticalSectionScoped cs(&_callbackCritSect);
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| +        rtc::CritScope cs(&_callbackCritSect);
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|          return _externalTransport;
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|      }
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|      bool ExternalMixing() const
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| @@ -460,7 +458,7 @@ public:
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|      // Used for obtaining RTT for a receive-only channel.
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|      void set_associate_send_channel(const ChannelOwner& channel) {
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|        assert(_channelId != channel.channel()->ChannelId());
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| -      CriticalSectionScoped lock(assoc_send_channel_lock_.get());
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| +      rtc::CritScope lock(&assoc_send_channel_lock_);
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|        associate_send_channel_ = channel;
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|      }
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|  
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| @@ -494,9 +492,9 @@ private:
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|      int32_t GetPlayoutFrequency();
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|      int64_t GetRTT(bool allow_associate_channel) const;
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|  
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| -    CriticalSectionWrapper& _fileCritSect;
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| -    CriticalSectionWrapper& _callbackCritSect;
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| -    CriticalSectionWrapper& volume_settings_critsect_;
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| +    mutable rtc::CriticalSection _fileCritSect;
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| +    mutable rtc::CriticalSection _callbackCritSect;
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| +    mutable rtc::CriticalSection volume_settings_critsect_;
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|      uint32_t _instanceId;
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|      int32_t _channelId;
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|  
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| @@ -544,7 +542,7 @@ private:
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|      uint16_t send_sequence_number_;
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|      uint8_t restored_packet_[kVoiceEngineMaxIpPacketSizeBytes];
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|  
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| -    rtc::scoped_ptr<CriticalSectionWrapper> ts_stats_lock_;
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| +    mutable rtc::CriticalSection ts_stats_lock_;
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|  
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|      rtc::scoped_ptr<rtc::TimestampWrapAroundHandler> rtp_ts_wraparound_handler_;
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|      // The rtp timestamp of the first played out audio frame.
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| @@ -560,7 +558,7 @@ private:
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|      ProcessThread* _moduleProcessThreadPtr;
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|      AudioDeviceModule* _audioDeviceModulePtr;
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|      VoiceEngineObserver* _voiceEngineObserverPtr; // owned by base
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| -    CriticalSectionWrapper* _callbackCritSectPtr; // owned by base
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| +    rtc::CriticalSection* _callbackCritSectPtr; // owned by base
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|      Transport* _transportPtr; // WebRtc socket or external transport
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|      RMSLevel rms_level_;
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|      rtc::scoped_ptr<AudioProcessing> rx_audioproc_;  // far end AudioProcessing
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| @@ -585,7 +583,7 @@ private:
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|      // VoENetwork
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|      AudioFrame::SpeechType _outputSpeechType;
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|      // VoEVideoSync
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| -    rtc::scoped_ptr<CriticalSectionWrapper> video_sync_lock_;
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| +    mutable rtc::CriticalSection video_sync_lock_;
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|      uint32_t _average_jitter_buffer_delay_us GUARDED_BY(video_sync_lock_);
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|      uint32_t _previousTimestamp;
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|      uint16_t _recPacketDelayMs GUARDED_BY(video_sync_lock_);
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| @@ -598,7 +596,7 @@ private:
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|      rtc::scoped_ptr<VoERtcpObserver> rtcp_observer_;
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|      rtc::scoped_ptr<NetworkPredictor> network_predictor_;
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|      // An associated send channel.
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| -    rtc::scoped_ptr<CriticalSectionWrapper> assoc_send_channel_lock_;
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| +    mutable rtc::CriticalSection assoc_send_channel_lock_;
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|      ChannelOwner associate_send_channel_ GUARDED_BY(assoc_send_channel_lock_);
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|  
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|      bool pacing_enabled_;
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| 
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