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