Index: webrtc/voice_engine/channel.h |
diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h |
index 889214b0aaf579edba1a44f13d64050263c51c3e..60c751e0d58f6408fe0a05e98748d78f2945a462 100644 |
--- a/webrtc/voice_engine/channel.h |
+++ b/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_; |